Hi All,
Sorry for cross-posting this email to PJLIB, but maybe there are some things related. Anyhow! I'm having problems on kamailio v4.0.2 under Fedora 18 64bit and TCP client like iPhone using PJSIP as SIP library. Basically once the iPhone side in close the call (TCP->UDP) I'm getting the error below. Kamailio is running under a VPS without NATed network so it uses a real public address. Furthermore, note that tcp_main is answering to a 192.168.2.98 ip address which is the iPhone client. This looks really strange to me since it should answer directly to the public/port used for the registration and not to a such kind of reserved address. The kamilio configuration is basically the default with a very few changes like NAT, rtpproxy and postgresql backend.
This problems doesn't happen at all when using UDP->UDP calls. But I cannot use it because as you certain know UDP connection under iPhone will not work when the application run in background mode.
Can someone suggest how to solve this issue or maybe suggest a TCP working solution for iPhone?
Thanks in advance. Roberto Fichera.
Jul 30 16:21:53 proxy /usr/sbin/kamailio[9502]: ERROR: <core> [tcp_main.c:4432]: tcpconn_main_timeout(): connect 192.168.2.98:5060 failed (timeout) Jul 30 16:21:55 proxy rtpproxy[2262]: DBUG:get_command: received command "9483_9 D 12d1d19926c4ff742a52f0c855b1bb83@94.94.x.x:5060 as74e0c388 GROahimCK6KTrl5CkYEg7nuoPIIXZ8cj" Jul 30 16:21:55 proxy rtpproxy[2262]: INFO:handle_delete: forcefully deleting session 1 on ports 15604/17354 Jul 30 16:21:55 proxy rtpproxy[2262]: INFO:remove_session: RTP stats: 354 in from callee, 603 in from caller, 957 relayed, 0 dropped Jul 30 16:21:55 proxy rtpproxy[2262]: INFO:remove_session: RTCP stats: 5 in from callee, 2 in from caller, 7 relayed, 0 dropped Jul 30 16:21:55 proxy rtpproxy[2262]: INFO:remove_session: session on ports 15604/17354 is cleaned up Jul 30 16:21:55 proxy rtpproxy[2262]: DBUG:doreply: sending reply "9483_9 0 Jul 30 16:21:55 proxy rtpproxy[2262]: " Jul 30 16:22:04 proxy /usr/sbin/kamailio[9502]: ERROR: <core> [tcp_main.c:4432]: tcpconn_main_timeout(): connect 192.168.2.98:5060 failed (timeout) Jul 30 16:22:14 proxy rtpproxy[2262]: DBUG:get_command: received command "9496_16 D 12d1d19926c4ff742a52f0c855b1bb83@94.94.x.x:5060 GROahimCK6KTrl5CkYEg7nuoPIIXZ8cj as74e0c388" Jul 30 16:22:14 proxy rtpproxy[2262]: INFO:handle_command: delete request failed: session 12d1d19926c4ff742a52f0c855b1bb83@94.94.x.x:5060, tags GROahimCK6KTrl5CkYEg7nuoPIIXZ8cj/as74e0c388 not found Jul 30 16:22:14 proxy rtpproxy[2262]: DBUG:doreply: sending reply "9496_16 E8 Jul 30 16:22:14 proxy rtpproxy[2262]: "
Hello,
you should grab the ngrep for such call to understand better what happens. Also, dumping the location records will be useful (kamctl ul show).
Also, be sure that tcp connection lifetime is long enough to survive re-registration. To avoid trying to open connections behind nat, use set_forward_no_connect() for calls involving nat traversal.
Cheers, Daniel
On 7/30/13 6:44 PM, Roberto Fichera wrote:
Hi All,
Sorry for cross-posting this email to PJLIB, but maybe there are some things related. Anyhow! I'm having problems on kamailio v4.0.2 under Fedora 18 64bit and TCP client like iPhone using PJSIP as SIP library. Basically once the iPhone side in close the call (TCP->UDP) I'm getting the error below. Kamailio is running under a VPS without NATed network so it uses a real public address. Furthermore, note that tcp_main is answering to a 192.168.2.98 ip address which is the iPhone client. This looks really strange to me since it should answer directly to the public/port used for the registration and not to a such kind of reserved address. The kamilio configuration is basically the default with a very few changes like NAT, rtpproxy and postgresql backend.
This problems doesn't happen at all when using UDP->UDP calls. But I cannot use it because as you certain know UDP connection under iPhone will not work when the application run in background mode.
Can someone suggest how to solve this issue or maybe suggest a TCP working solution for iPhone?
Thanks in advance. Roberto Fichera.
Jul 30 16:21:53 proxy /usr/sbin/kamailio[9502]: ERROR: <core> [tcp_main.c:4432]: tcpconn_main_timeout(): connect 192.168.2.98:5060 failed (timeout) Jul 30 16:21:55 proxy rtpproxy[2262]: DBUG:get_command: received command "9483_9 D 12d1d19926c4ff742a52f0c855b1bb83@94.94.x.x:5060 as74e0c388 GROahimCK6KTrl5CkYEg7nuoPIIXZ8cj" Jul 30 16:21:55 proxy rtpproxy[2262]: INFO:handle_delete: forcefully deleting session 1 on ports 15604/17354 Jul 30 16:21:55 proxy rtpproxy[2262]: INFO:remove_session: RTP stats: 354 in from callee, 603 in from caller, 957 relayed, 0 dropped Jul 30 16:21:55 proxy rtpproxy[2262]: INFO:remove_session: RTCP stats: 5 in from callee, 2 in from caller, 7 relayed, 0 dropped Jul 30 16:21:55 proxy rtpproxy[2262]: INFO:remove_session: session on ports 15604/17354 is cleaned up Jul 30 16:21:55 proxy rtpproxy[2262]: DBUG:doreply: sending reply "9483_9 0 Jul 30 16:21:55 proxy rtpproxy[2262]: " Jul 30 16:22:04 proxy /usr/sbin/kamailio[9502]: ERROR: <core> [tcp_main.c:4432]: tcpconn_main_timeout(): connect 192.168.2.98:5060 failed (timeout) Jul 30 16:22:14 proxy rtpproxy[2262]: DBUG:get_command: received command "9496_16 D 12d1d19926c4ff742a52f0c855b1bb83@94.94.x.x:5060 GROahimCK6KTrl5CkYEg7nuoPIIXZ8cj as74e0c388" Jul 30 16:22:14 proxy rtpproxy[2262]: INFO:handle_command: delete request failed: session 12d1d19926c4ff742a52f0c855b1bb83@94.94.x.x:5060, tags GROahimCK6KTrl5CkYEg7nuoPIIXZ8cj/as74e0c388 not found Jul 30 16:22:14 proxy rtpproxy[2262]: DBUG:doreply: sending reply "9496_16 E8 Jul 30 16:22:14 proxy rtpproxy[2262]: "
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
On 08/13/2013 12:03 PM, Daniel-Constantin Mierla wrote:
Hello,
you should grab the ngrep for such call to understand better what happens. Also, dumping the location records will be useful (kamctl ul show).
Also, be sure that tcp connection lifetime is long enough to survive re-registration. To avoid trying to open connections behind nat, use set_forward_no_connect() for calls involving nat traversal.
I'm using the default conf coming from fedora rpm. So, mainly the problem seems related to kamailio which doesn't reuse the TCP port used by NATed clients. I've also notice that the received field isn't set at all, so this means that the contact will not get aliased at all.
I would really like to have a look to a working cfg file for TCP NATed clients that reuse the TCP port. Even better if the configuration is based on the fedora default rpm.
Cheers, Roberto Fichera.
Cheers, Daniel
On 7/30/13 6:44 PM, Roberto Fichera wrote:
Hi All,
Sorry for cross-posting this email to PJLIB, but maybe there are some things related. Anyhow! I'm having problems on kamailio v4.0.2 under Fedora 18 64bit and TCP client like iPhone using PJSIP as SIP library. Basically once the iPhone side in close the call (TCP->UDP) I'm getting the error below. Kamailio is running under a VPS without NATed network so it uses a real public address. Furthermore, note that tcp_main is answering to a 192.168.2.98 ip address which is the iPhone client. This looks really strange to me since it should answer directly to the public/port used for the registration and not to a such kind of reserved address. The kamilio configuration is basically the default with a very few changes like NAT, rtpproxy and postgresql backend.
This problems doesn't happen at all when using UDP->UDP calls. But I cannot use it because as you certain know UDP connection under iPhone will not work when the application run in background mode.
Can someone suggest how to solve this issue or maybe suggest a TCP working solution for iPhone?
Thanks in advance. Roberto Fichera.
Jul 30 16:21:53 proxy /usr/sbin/kamailio[9502]: ERROR: <core> [tcp_main.c:4432]: tcpconn_main_timeout(): connect 192.168.2.98:5060 failed (timeout) Jul 30 16:21:55 proxy rtpproxy[2262]: DBUG:get_command: received command "9483_9 D 12d1d19926c4ff742a52f0c855b1bb83@94.94.x.x:5060 as74e0c388 GROahimCK6KTrl5CkYEg7nuoPIIXZ8cj" Jul 30 16:21:55 proxy rtpproxy[2262]: INFO:handle_delete: forcefully deleting session 1 on ports 15604/17354 Jul 30 16:21:55 proxy rtpproxy[2262]: INFO:remove_session: RTP stats: 354 in from callee, 603 in from caller, 957 relayed, 0 dropped Jul 30 16:21:55 proxy rtpproxy[2262]: INFO:remove_session: RTCP stats: 5 in from callee, 2 in from caller, 7 relayed, 0 dropped Jul 30 16:21:55 proxy rtpproxy[2262]: INFO:remove_session: session on ports 15604/17354 is cleaned up Jul 30 16:21:55 proxy rtpproxy[2262]: DBUG:doreply: sending reply "9483_9 0 Jul 30 16:21:55 proxy rtpproxy[2262]: " Jul 30 16:22:04 proxy /usr/sbin/kamailio[9502]: ERROR: <core> [tcp_main.c:4432]: tcpconn_main_timeout(): connect 192.168.2.98:5060 failed (timeout) Jul 30 16:22:14 proxy rtpproxy[2262]: DBUG:get_command: received command "9496_16 D 12d1d19926c4ff742a52f0c855b1bb83@94.94.x.x:5060 GROahimCK6KTrl5CkYEg7nuoPIIXZ8cj as74e0c388" Jul 30 16:22:14 proxy rtpproxy[2262]: INFO:handle_command: delete request failed: session 12d1d19926c4ff742a52f0c855b1bb83@94.94.x.x:5060, tags GROahimCK6KTrl5CkYEg7nuoPIIXZ8cj/as74e0c388 not found Jul 30 16:22:14 proxy rtpproxy[2262]: DBUG:doreply: sending reply "9496_16 E8 Jul 30 16:22:14 proxy rtpproxy[2262]: "
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Hello,
On 8/13/13 1:10 PM, Roberto Fichera wrote:
On 08/13/2013 12:03 PM, Daniel-Constantin Mierla wrote:
Hello,
you should grab the ngrep for such call to understand better what happens. Also, dumping the location records will be useful (kamctl ul show).
Also, be sure that tcp connection lifetime is long enough to survive re-registration. To avoid trying to open connections behind nat, use set_forward_no_connect() for calls involving nat traversal.
I'm using the default conf coming from fedora rpm. So, mainly the problem seems related to kamailio which doesn't reuse the TCP port used by NATed clients. I've also notice that the received field isn't set at all, so this means that the contact will not get aliased at all.
I would really like to have a look to a working cfg file for TCP NATed clients that reuse the TCP port. Even better if the configuration is based on the fedora default rpm.
if received is not set, then means the register was not detected as coming from behind nat. Is the phone using stun?
Again, put here the ngrep for registration and a call to see if something is wrong with signaling. There is no help that we can provide otherwise. The default config works fine for tcp and natted clients, I use it everywhere for this case without issues.
Cheers, Daniel
Cheers, Roberto Fichera.
Cheers, Daniel
On 7/30/13 6:44 PM, Roberto Fichera wrote:
Hi All,
Sorry for cross-posting this email to PJLIB, but maybe there are some things related. Anyhow! I'm having problems on kamailio v4.0.2 under Fedora 18 64bit and TCP client like iPhone using PJSIP as SIP library. Basically once the iPhone side in close the call (TCP->UDP) I'm getting the error below. Kamailio is running under a VPS without NATed network so it uses a real public address. Furthermore, note that tcp_main is answering to a 192.168.2.98 ip address which is the iPhone client. This looks really strange to me since it should answer directly to the public/port used for the registration and not to a such kind of reserved address. The kamilio configuration is basically the default with a very few changes like NAT, rtpproxy and postgresql backend.
This problems doesn't happen at all when using UDP->UDP calls. But I cannot use it because as you certain know UDP connection under iPhone will not work when the application run in background mode.
Can someone suggest how to solve this issue or maybe suggest a TCP working solution for iPhone?
Thanks in advance. Roberto Fichera.
Jul 30 16:21:53 proxy /usr/sbin/kamailio[9502]: ERROR: <core> [tcp_main.c:4432]: tcpconn_main_timeout(): connect 192.168.2.98:5060 failed (timeout) Jul 30 16:21:55 proxy rtpproxy[2262]: DBUG:get_command: received command "9483_9 D 12d1d19926c4ff742a52f0c855b1bb83@94.94.x.x:5060 as74e0c388 GROahimCK6KTrl5CkYEg7nuoPIIXZ8cj" Jul 30 16:21:55 proxy rtpproxy[2262]: INFO:handle_delete: forcefully deleting session 1 on ports 15604/17354 Jul 30 16:21:55 proxy rtpproxy[2262]: INFO:remove_session: RTP stats: 354 in from callee, 603 in from caller, 957 relayed, 0 dropped Jul 30 16:21:55 proxy rtpproxy[2262]: INFO:remove_session: RTCP stats: 5 in from callee, 2 in from caller, 7 relayed, 0 dropped Jul 30 16:21:55 proxy rtpproxy[2262]: INFO:remove_session: session on ports 15604/17354 is cleaned up Jul 30 16:21:55 proxy rtpproxy[2262]: DBUG:doreply: sending reply "9483_9 0 Jul 30 16:21:55 proxy rtpproxy[2262]: " Jul 30 16:22:04 proxy /usr/sbin/kamailio[9502]: ERROR: <core> [tcp_main.c:4432]: tcpconn_main_timeout(): connect 192.168.2.98:5060 failed (timeout) Jul 30 16:22:14 proxy rtpproxy[2262]: DBUG:get_command: received command "9496_16 D 12d1d19926c4ff742a52f0c855b1bb83@94.94.x.x:5060 GROahimCK6KTrl5CkYEg7nuoPIIXZ8cj as74e0c388" Jul 30 16:22:14 proxy rtpproxy[2262]: INFO:handle_command: delete request failed: session 12d1d19926c4ff742a52f0c855b1bb83@94.94.x.x:5060, tags GROahimCK6KTrl5CkYEg7nuoPIIXZ8cj/as74e0c388 not found Jul 30 16:22:14 proxy rtpproxy[2262]: DBUG:doreply: sending reply "9496_16 E8 Jul 30 16:22:14 proxy rtpproxy[2262]: "
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
On 08/13/2013 02:33 PM, Daniel-Constantin Mierla wrote:
Hello,
On 8/13/13 1:10 PM, Roberto Fichera wrote:
On 08/13/2013 12:03 PM, Daniel-Constantin Mierla wrote:
Hello,
you should grab the ngrep for such call to understand better what happens. Also, dumping the location records will be useful (kamctl ul show).
Also, be sure that tcp connection lifetime is long enough to survive re-registration. To avoid trying to open connections behind nat, use set_forward_no_connect() for calls involving nat traversal.
I'm using the default conf coming from fedora rpm. So, mainly the problem seems related to kamailio which doesn't reuse the TCP port used by NATed clients. I've also notice that the received field isn't set at all, so this means that the contact will not get aliased at all.
I would really like to have a look to a working cfg file for TCP NATed clients that reuse the TCP port. Even better if the configuration is based on the fedora default rpm.
if received is not set, then means the register was not detected as coming from behind nat. Is the phone using stun?
I'm testing with a normal rtpproxy configuration. BTW udp -> udp work perfectly.
Again, put here the ngrep for registration and a call to see if something is wrong with signaling. There is no help that we can provide otherwise. The default config works fine for tcp and natted clients, I use it everywhere for this case without issues.
I tried the default cfg enabling both NAT and RTPproxy, but seems that kamailio doesn't reuse TCP ports. Anyway, this is a call from UDP (512) -> TCP (526) both behind the same NAT, from kamailio point of view
[root@proxy ~]# kamctl ul show 526 Contact:: sip:526@94.94.X.X:1238;transport=TCP;ob;q=;expires=537;flags=0x0;cflags=0x40;state=0;socket=tcp:178.79.x.x:5060;methods=0x1FDF;received=sip:94.94.X.X:61922;transport=TCP;user_agent=<DICE Smartphone 1.0/iPhone>;reg-id=0 [root@proxy ~]# kamctl ul show 512 Contact:: sip:512@94.94.X.X:5060;q=;expires=32;flags=0x0;cflags=0x40;state=0;socket=udp:178.79.x.x:5060;methods=0xFFFFFFFF;received=sip:94.94.X.X:1025;user_agent=<DICE 1.8.20.1>;reg-id=0 [root@proxy ~]#
U 94.94.X.X:1025 -> 178.79.x.x:5060 INVITE sip:526@test.domain:5060 SIP/2.0. Via: SIP/2.0/UDP 94.94.X.X:5060;branch=z9hG4bK4a420cee;rport. Max-Forwards: 70. From: "asterisk" sip:512@test.domain;tag=as76007db0. To: sip:526@test.domain:5060. Contact: sip:512@94.94.X.X:5060. Call-ID: 068a5a23639785a7583d952d6f9bca84@test.domain. CSeq: 102 INVITE. User-Agent: DICE 1.8.20.1. Date: Tue, 13 Aug 2013 13:04:30 GMT. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH. Supported: replaces, timer. Content-Type: application/sdp. Content-Length: 264. . v=0. o=root 1263161426 1263161426 IN IP4 94.94.X.X. s=Asterisk PBX 11.3.0. c=IN IP4 94.94.X.X. t=0 0. m=audio 10782 RTP/AVP 0 110 101. a=rtpmap:0 PCMU/8000. a=rtpmap:110 speex/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=sendrecv.
# U 178.79.x.x:5060 -> 94.94.X.X:1025 SIP/2.0 407 Proxy Authentication Required. Via: SIP/2.0/UDP 94.94.X.X:5060;branch=z9hG4bK4a420cee;rport=1025. From: "asterisk" sip:512@test.domain;tag=as76007db0. To: sip:526@test.domain:5060;tag=333a0370df4a40d5d5a0c21bb156e2a6.4a00. Call-ID: 068a5a23639785a7583d952d6f9bca84@test.domain. CSeq: 102 INVITE. Proxy-Authenticate: Digest realm="test.domain", nonce="UgowYFIKLzQXDSUWUpy4xlhdXZ7WNjPt". Server: kamailio (4.0.2 (x86_64/linux)). Content-Length: 0. .
# U 94.94.X.X:1025 -> 178.79.x.x:5060 INVITE sip:526@test.domain:5060 SIP/2.0. Via: SIP/2.0/UDP 94.94.X.X:5060;branch=z9hG4bK4a420cee;rport. Max-Forwards: 70. From: "asterisk" sip:512@test.domain;tag=as76007db0. To: sip:526@test.domain:5060. Contact: sip:512@94.94.X.X:5060. Call-ID: 068a5a23639785a7583d952d6f9bca84@test.domain. CSeq: 102 INVITE. User-Agent: DICE 1.8.20.1. Date: Tue, 13 Aug 2013 13:04:30 GMT. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH. Supported: replaces, timer. Content-Type: application/sdp. Content-Length: 264. . v=0. o=root 1263161426 1263161426 IN IP4 94.94.X.X. s=Asterisk PBX 11.3.0. c=IN IP4 94.94.X.X. t=0 0. m=audio 10782 RTP/AVP 0 110 101. a=rtpmap:0 PCMU/8000. a=rtpmap:110 speex/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=sendrecv.
# U 178.79.x.x:5060 -> 94.94.X.X:1025 SIP/2.0 407 Proxy Authentication Required. Via: SIP/2.0/UDP 94.94.X.X:5060;branch=z9hG4bK4a420cee;rport=1025. From: "asterisk" sip:512@test.domain;tag=as76007db0. To: sip:526@test.domain:5060;tag=333a0370df4a40d5d5a0c21bb156e2a6.4a00. Call-ID: 068a5a23639785a7583d952d6f9bca84@test.domain. CSeq: 102 INVITE. Proxy-Authenticate: Digest realm="test.domain", nonce="UgowYFIKLzQXDSUWUpy4xlhdXZ7WNjPt". Server: kamailio (4.0.2 (x86_64/linux)). Content-Length: 0. .
# U 94.94.X.X:1025 -> 178.79.x.x:5060 ACK sip:526@test.domain:5060 SIP/2.0. Via: SIP/2.0/UDP 94.94.X.X:5060;branch=z9hG4bK4a420cee;rport. Max-Forwards: 70. From: "asterisk" sip:512@test.domain;tag=as76007db0. To: sip:526@test.domain:5060;tag=333a0370df4a40d5d5a0c21bb156e2a6.4a00. Contact: sip:512@94.94.X.X:5060. Call-ID: 068a5a23639785a7583d952d6f9bca84@test.domain. CSeq: 102 ACK. User-Agent: DICE 1.8.20.1. Content-Length: 0. .
# U 94.94.X.X:1025 -> 178.79.x.x:5060 ACK sip:526@test.domain:5060 SIP/2.0. Via: SIP/2.0/UDP 94.94.X.X:5060;branch=z9hG4bK4a420cee;rport. Max-Forwards: 70. From: "asterisk" sip:512@test.domain;tag=as76007db0. To: sip:526@test.domain:5060. Contact: sip:512@94.94.X.X:5060. Call-ID: 068a5a23639785a7583d952d6f9bca84@test.domain. CSeq: 102 ACK. User-Agent: DICE 1.8.20.1. Content-Length: 0. .
# T 178.79.x.x:5060 -> 94.94.X.X:61922 [AP] ACK sip:526@94.94.X.X:1238;transport=TCP;ob SIP/2.0. Via: SIP/2.0/TCP 178.79.x.x;branch=z9hG4bKcydzigwkX. Via: SIP/2.0/UDP 94.94.X.X:5060;branch=z9hG4bK4a420cee;rport=1025. Max-Forwards: 16. From: "asterisk" sip:512@test.domain;tag=as76007db0. To: sip:526@test.domain:5060. Contact: sip:512@94.94.X.X:1025. Call-ID: 068a5a23639785a7583d952d6f9bca84@test.domain. CSeq: 102 ACK. User-Agent: DICE 1.8.20.1. Content-Length: 0. .
# T 178.79.x.x:5060 -> 94.94.X.X:61922 [AP] ACK sip:526@94.94.X.X:1238;transport=TCP;ob SIP/2.0. Via: SIP/2.0/TCP 178.79.x.x;branch=z9hG4bKcydzigwkX. Via: SIP/2.0/UDP 94.94.X.X:5060;branch=z9hG4bK4a420cee;rport=1025. Max-Forwards: 16. From: "asterisk" sip:512@test.domain;tag=as76007db0. To: sip:526@test.domain:5060. Contact: sip:512@94.94.X.X:1025. Call-ID: 068a5a23639785a7583d952d6f9bca84@test.domain. CSeq: 102 ACK. User-Agent: DICE 1.8.20.1. Content-Length: 0. .
Cheers, Daniel
Cheers, Roberto Fichera.
Cheers, Daniel
On 7/30/13 6:44 PM, Roberto Fichera wrote:
Hi All,
Sorry for cross-posting this email to PJLIB, but maybe there are some things related. Anyhow! I'm having problems on kamailio v4.0.2 under Fedora 18 64bit and TCP client like iPhone using PJSIP as SIP library. Basically once the iPhone side in close the call (TCP->UDP) I'm getting the error below. Kamailio is running under a VPS without NATed network so it uses a real public address. Furthermore, note that tcp_main is answering to a 192.168.2.98 ip address which is the iPhone client. This looks really strange to me since it should answer directly to the public/port used for the registration and not to a such kind of reserved address. The kamilio configuration is basically the default with a very few changes like NAT, rtpproxy and postgresql backend.
This problems doesn't happen at all when using UDP->UDP calls. But I cannot use it because as you certain know UDP connection under iPhone will not work when the application run in background mode.
Can someone suggest how to solve this issue or maybe suggest a TCP working solution for iPhone?
Thanks in advance. Roberto Fichera.
Jul 30 16:21:53 proxy /usr/sbin/kamailio[9502]: ERROR: <core> [tcp_main.c:4432]: tcpconn_main_timeout(): connect 192.168.2.98:5060 failed (timeout) Jul 30 16:21:55 proxy rtpproxy[2262]: DBUG:get_command: received command "9483_9 D 12d1d19926c4ff742a52f0c855b1bb83@94.94.x.x:5060 as74e0c388 GROahimCK6KTrl5CkYEg7nuoPIIXZ8cj" Jul 30 16:21:55 proxy rtpproxy[2262]: INFO:handle_delete: forcefully deleting session 1 on ports 15604/17354 Jul 30 16:21:55 proxy rtpproxy[2262]: INFO:remove_session: RTP stats: 354 in from callee, 603 in from caller, 957 relayed, 0 dropped Jul 30 16:21:55 proxy rtpproxy[2262]: INFO:remove_session: RTCP stats: 5 in from callee, 2 in from caller, 7 relayed, 0 dropped Jul 30 16:21:55 proxy rtpproxy[2262]: INFO:remove_session: session on ports 15604/17354 is cleaned up Jul 30 16:21:55 proxy rtpproxy[2262]: DBUG:doreply: sending reply "9483_9 0 Jul 30 16:21:55 proxy rtpproxy[2262]: " Jul 30 16:22:04 proxy /usr/sbin/kamailio[9502]: ERROR: <core> [tcp_main.c:4432]: tcpconn_main_timeout(): connect 192.168.2.98:5060 failed (timeout) Jul 30 16:22:14 proxy rtpproxy[2262]: DBUG:get_command: received command "9496_16 D 12d1d19926c4ff742a52f0c855b1bb83@94.94.x.x:5060 GROahimCK6KTrl5CkYEg7nuoPIIXZ8cj as74e0c388" Jul 30 16:22:14 proxy rtpproxy[2262]: INFO:handle_command: delete request failed: session 12d1d19926c4ff742a52f0c855b1bb83@94.94.x.x:5060, tags GROahimCK6KTrl5CkYEg7nuoPIIXZ8cj/as74e0c388 not found Jul 30 16:22:14 proxy rtpproxy[2262]: DBUG:doreply: sending reply "9496_16 E8 Jul 30 16:22:14 proxy rtpproxy[2262]: "
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
The sip trace is incomplete, I don't see invite with credentials and then I see an ACK sent via tcp on the right connection. That means a negative response was received over tcp from the callee.
Daniel
On 8/13/13 3:15 PM, Roberto Fichera wrote:
On 08/13/2013 02:33 PM, Daniel-Constantin Mierla wrote:
Hello,
On 8/13/13 1:10 PM, Roberto Fichera wrote:
On 08/13/2013 12:03 PM, Daniel-Constantin Mierla wrote:
Hello,
you should grab the ngrep for such call to understand better what happens. Also, dumping the location records will be useful (kamctl ul show).
Also, be sure that tcp connection lifetime is long enough to survive re-registration. To avoid trying to open connections behind nat, use set_forward_no_connect() for calls involving nat traversal.
I'm using the default conf coming from fedora rpm. So, mainly the problem seems related to kamailio which doesn't reuse the TCP port used by NATed clients. I've also notice that the received field isn't set at all, so this means that the contact will not get aliased at all.
I would really like to have a look to a working cfg file for TCP NATed clients that reuse the TCP port. Even better if the configuration is based on the fedora default rpm.
if received is not set, then means the register was not detected as coming from behind nat. Is the phone using stun?
I'm testing with a normal rtpproxy configuration. BTW udp -> udp work perfectly.
Again, put here the ngrep for registration and a call to see if something is wrong with signaling. There is no help that we can provide otherwise. The default config works fine for tcp and natted clients, I use it everywhere for this case without issues.
I tried the default cfg enabling both NAT and RTPproxy, but seems that kamailio doesn't reuse TCP ports. Anyway, this is a call from UDP (512) -> TCP (526) both behind the same NAT, from kamailio point of view
[root@proxy ~]# kamctl ul show 526 Contact:: sip:526@94.94.X.X:1238;transport=TCP;ob;q=;expires=537;flags=0x0;cflags=0x40;state=0;socket=tcp:178.79.x.x:5060;methods=0x1FDF;received=sip:94.94.X.X:61922;transport=TCP;user_agent=<DICE Smartphone 1.0/iPhone>;reg-id=0 [root@proxy ~]# kamctl ul show 512 Contact:: sip:512@94.94.X.X:5060;q=;expires=32;flags=0x0;cflags=0x40;state=0;socket=udp:178.79.x.x:5060;methods=0xFFFFFFFF;received=sip:94.94.X.X:1025;user_agent=<DICE 1.8.20.1>;reg-id=0 [root@proxy ~]#
U 94.94.X.X:1025 -> 178.79.x.x:5060 INVITE sip:526@test.domain:5060 SIP/2.0. Via: SIP/2.0/UDP 94.94.X.X:5060;branch=z9hG4bK4a420cee;rport. Max-Forwards: 70. From: "asterisk" sip:512@test.domain;tag=as76007db0. To: sip:526@test.domain:5060. Contact: sip:512@94.94.X.X:5060. Call-ID: 068a5a23639785a7583d952d6f9bca84@test.domain. CSeq: 102 INVITE. User-Agent: DICE 1.8.20.1. Date: Tue, 13 Aug 2013 13:04:30 GMT. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH. Supported: replaces, timer. Content-Type: application/sdp. Content-Length: 264. . v=0. o=root 1263161426 1263161426 IN IP4 94.94.X.X. s=Asterisk PBX 11.3.0. c=IN IP4 94.94.X.X. t=0 0. m=audio 10782 RTP/AVP 0 110 101. a=rtpmap:0 PCMU/8000. a=rtpmap:110 speex/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=sendrecv.
# U 178.79.x.x:5060 -> 94.94.X.X:1025 SIP/2.0 407 Proxy Authentication Required. Via: SIP/2.0/UDP 94.94.X.X:5060;branch=z9hG4bK4a420cee;rport=1025. From: "asterisk" sip:512@test.domain;tag=as76007db0. To: sip:526@test.domain:5060;tag=333a0370df4a40d5d5a0c21bb156e2a6.4a00. Call-ID: 068a5a23639785a7583d952d6f9bca84@test.domain. CSeq: 102 INVITE. Proxy-Authenticate: Digest realm="test.domain", nonce="UgowYFIKLzQXDSUWUpy4xlhdXZ7WNjPt". Server: kamailio (4.0.2 (x86_64/linux)). Content-Length: 0. .
# U 94.94.X.X:1025 -> 178.79.x.x:5060 INVITE sip:526@test.domain:5060 SIP/2.0. Via: SIP/2.0/UDP 94.94.X.X:5060;branch=z9hG4bK4a420cee;rport. Max-Forwards: 70. From: "asterisk" sip:512@test.domain;tag=as76007db0. To: sip:526@test.domain:5060. Contact: sip:512@94.94.X.X:5060. Call-ID: 068a5a23639785a7583d952d6f9bca84@test.domain. CSeq: 102 INVITE. User-Agent: DICE 1.8.20.1. Date: Tue, 13 Aug 2013 13:04:30 GMT. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH. Supported: replaces, timer. Content-Type: application/sdp. Content-Length: 264. . v=0. o=root 1263161426 1263161426 IN IP4 94.94.X.X. s=Asterisk PBX 11.3.0. c=IN IP4 94.94.X.X. t=0 0. m=audio 10782 RTP/AVP 0 110 101. a=rtpmap:0 PCMU/8000. a=rtpmap:110 speex/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=sendrecv.
# U 178.79.x.x:5060 -> 94.94.X.X:1025 SIP/2.0 407 Proxy Authentication Required. Via: SIP/2.0/UDP 94.94.X.X:5060;branch=z9hG4bK4a420cee;rport=1025. From: "asterisk" sip:512@test.domain;tag=as76007db0. To: sip:526@test.domain:5060;tag=333a0370df4a40d5d5a0c21bb156e2a6.4a00. Call-ID: 068a5a23639785a7583d952d6f9bca84@test.domain. CSeq: 102 INVITE. Proxy-Authenticate: Digest realm="test.domain", nonce="UgowYFIKLzQXDSUWUpy4xlhdXZ7WNjPt". Server: kamailio (4.0.2 (x86_64/linux)). Content-Length: 0. .
# U 94.94.X.X:1025 -> 178.79.x.x:5060 ACK sip:526@test.domain:5060 SIP/2.0. Via: SIP/2.0/UDP 94.94.X.X:5060;branch=z9hG4bK4a420cee;rport. Max-Forwards: 70. From: "asterisk" sip:512@test.domain;tag=as76007db0. To: sip:526@test.domain:5060;tag=333a0370df4a40d5d5a0c21bb156e2a6.4a00. Contact: sip:512@94.94.X.X:5060. Call-ID: 068a5a23639785a7583d952d6f9bca84@test.domain. CSeq: 102 ACK. User-Agent: DICE 1.8.20.1. Content-Length: 0. .
# U 94.94.X.X:1025 -> 178.79.x.x:5060 ACK sip:526@test.domain:5060 SIP/2.0. Via: SIP/2.0/UDP 94.94.X.X:5060;branch=z9hG4bK4a420cee;rport. Max-Forwards: 70. From: "asterisk" sip:512@test.domain;tag=as76007db0. To: sip:526@test.domain:5060. Contact: sip:512@94.94.X.X:5060. Call-ID: 068a5a23639785a7583d952d6f9bca84@test.domain. CSeq: 102 ACK. User-Agent: DICE 1.8.20.1. Content-Length: 0. .
# T 178.79.x.x:5060 -> 94.94.X.X:61922 [AP] ACK sip:526@94.94.X.X:1238;transport=TCP;ob SIP/2.0. Via: SIP/2.0/TCP 178.79.x.x;branch=z9hG4bKcydzigwkX. Via: SIP/2.0/UDP 94.94.X.X:5060;branch=z9hG4bK4a420cee;rport=1025. Max-Forwards: 16. From: "asterisk" sip:512@test.domain;tag=as76007db0. To: sip:526@test.domain:5060. Contact: sip:512@94.94.X.X:1025. Call-ID: 068a5a23639785a7583d952d6f9bca84@test.domain. CSeq: 102 ACK. User-Agent: DICE 1.8.20.1. Content-Length: 0. .
# T 178.79.x.x:5060 -> 94.94.X.X:61922 [AP] ACK sip:526@94.94.X.X:1238;transport=TCP;ob SIP/2.0. Via: SIP/2.0/TCP 178.79.x.x;branch=z9hG4bKcydzigwkX. Via: SIP/2.0/UDP 94.94.X.X:5060;branch=z9hG4bK4a420cee;rport=1025. Max-Forwards: 16. From: "asterisk" sip:512@test.domain;tag=as76007db0. To: sip:526@test.domain:5060. Contact: sip:512@94.94.X.X:1025. Call-ID: 068a5a23639785a7583d952d6f9bca84@test.domain. CSeq: 102 ACK. User-Agent: DICE 1.8.20.1. Content-Length: 0. .
Cheers, Daniel
Cheers, Roberto Fichera.
Cheers, Daniel
On 7/30/13 6:44 PM, Roberto Fichera wrote:
Hi All,
Sorry for cross-posting this email to PJLIB, but maybe there are some things related. Anyhow! I'm having problems on kamailio v4.0.2 under Fedora 18 64bit and TCP client like iPhone using PJSIP as SIP library. Basically once the iPhone side in close the call (TCP->UDP) I'm getting the error below. Kamailio is running under a VPS without NATed network so it uses a real public address. Furthermore, note that tcp_main is answering to a 192.168.2.98 ip address which is the iPhone client. This looks really strange to me since it should answer directly to the public/port used for the registration and not to a such kind of reserved address. The kamilio configuration is basically the default with a very few changes like NAT, rtpproxy and postgresql backend.
This problems doesn't happen at all when using UDP->UDP calls. But I cannot use it because as you certain know UDP connection under iPhone will not work when the application run in background mode.
Can someone suggest how to solve this issue or maybe suggest a TCP working solution for iPhone?
Thanks in advance. Roberto Fichera.
Jul 30 16:21:53 proxy /usr/sbin/kamailio[9502]: ERROR: <core> [tcp_main.c:4432]: tcpconn_main_timeout(): connect 192.168.2.98:5060 failed (timeout) Jul 30 16:21:55 proxy rtpproxy[2262]: DBUG:get_command: received command "9483_9 D 12d1d19926c4ff742a52f0c855b1bb83@94.94.x.x:5060 as74e0c388 GROahimCK6KTrl5CkYEg7nuoPIIXZ8cj" Jul 30 16:21:55 proxy rtpproxy[2262]: INFO:handle_delete: forcefully deleting session 1 on ports 15604/17354 Jul 30 16:21:55 proxy rtpproxy[2262]: INFO:remove_session: RTP stats: 354 in from callee, 603 in from caller, 957 relayed, 0 dropped Jul 30 16:21:55 proxy rtpproxy[2262]: INFO:remove_session: RTCP stats: 5 in from callee, 2 in from caller, 7 relayed, 0 dropped Jul 30 16:21:55 proxy rtpproxy[2262]: INFO:remove_session: session on ports 15604/17354 is cleaned up Jul 30 16:21:55 proxy rtpproxy[2262]: DBUG:doreply: sending reply "9483_9 0 Jul 30 16:21:55 proxy rtpproxy[2262]: " Jul 30 16:22:04 proxy /usr/sbin/kamailio[9502]: ERROR: <core> [tcp_main.c:4432]: tcpconn_main_timeout(): connect 192.168.2.98:5060 failed (timeout) Jul 30 16:22:14 proxy rtpproxy[2262]: DBUG:get_command: received command "9496_16 D 12d1d19926c4ff742a52f0c855b1bb83@94.94.x.x:5060 GROahimCK6KTrl5CkYEg7nuoPIIXZ8cj as74e0c388" Jul 30 16:22:14 proxy rtpproxy[2262]: INFO:handle_command: delete request failed: session 12d1d19926c4ff742a52f0c855b1bb83@94.94.x.x:5060, tags GROahimCK6KTrl5CkYEg7nuoPIIXZ8cj/as74e0c388 not found Jul 30 16:22:14 proxy rtpproxy[2262]: DBUG:doreply: sending reply "9496_16 E8 Jul 30 16:22:14 proxy rtpproxy[2262]: "
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
On 08/13/2013 03:22 PM, Daniel-Constantin Mierla wrote:
The sip trace is incomplete, I don't see invite with credentials and then I see an ACK sent via tcp on the right connection. That means a negative response was received over tcp from the callee.
I think I'll restart from the default configuration so I'll removing all testing code I wrote ;-) !
Daniel
On 8/13/13 3:15 PM, Roberto Fichera wrote:
On 08/13/2013 02:33 PM, Daniel-Constantin Mierla wrote:
Hello,
On 8/13/13 1:10 PM, Roberto Fichera wrote:
On 08/13/2013 12:03 PM, Daniel-Constantin Mierla wrote:
Hello,
you should grab the ngrep for such call to understand better what happens. Also, dumping the location records will be useful (kamctl ul show).
Also, be sure that tcp connection lifetime is long enough to survive re-registration. To avoid trying to open connections behind nat, use set_forward_no_connect() for calls involving nat traversal.
I'm using the default conf coming from fedora rpm. So, mainly the problem seems related to kamailio which doesn't reuse the TCP port used by NATed clients. I've also notice that the received field isn't set at all, so this means that the contact will not get aliased at all.
I would really like to have a look to a working cfg file for TCP NATed clients that reuse the TCP port. Even better if the configuration is based on the fedora default rpm.
if received is not set, then means the register was not detected as coming from behind nat. Is the phone using stun?
I'm testing with a normal rtpproxy configuration. BTW udp -> udp work perfectly.
Again, put here the ngrep for registration and a call to see if something is wrong with signaling. There is no help that we can provide otherwise. The default config works fine for tcp and natted clients, I use it everywhere for this case without issues.
I tried the default cfg enabling both NAT and RTPproxy, but seems that kamailio doesn't reuse TCP ports. Anyway, this is a call from UDP (512) -> TCP (526) both behind the same NAT, from kamailio point of view
[root@proxy ~]# kamctl ul show 526 Contact:: sip:526@94.94.X.X:1238;transport=TCP;ob;q=;expires=537;flags=0x0;cflags=0x40;state=0;socket=tcp:178.79.x.x:5060;methods=0x1FDF;received=sip:94.94.X.X:61922;transport=TCP;user_agent=<DICE
Smartphone 1.0/iPhone>;reg-id=0 [root@proxy ~]# kamctl ul show 512 Contact:: sip:512@94.94.X.X:5060;q=;expires=32;flags=0x0;cflags=0x40;state=0;socket=udp:178.79.x.x:5060;methods=0xFFFFFFFF;received=sip:94.94.X.X:1025;user_agent=<DICE
1.8.20.1>;reg-id=0 [root@proxy ~]#
U 94.94.X.X:1025 -> 178.79.x.x:5060 INVITE sip:526@test.domain:5060 SIP/2.0. Via: SIP/2.0/UDP 94.94.X.X:5060;branch=z9hG4bK4a420cee;rport. Max-Forwards: 70. From: "asterisk" sip:512@test.domain;tag=as76007db0. To: sip:526@test.domain:5060. Contact: sip:512@94.94.X.X:5060. Call-ID: 068a5a23639785a7583d952d6f9bca84@test.domain. CSeq: 102 INVITE. User-Agent: DICE 1.8.20.1. Date: Tue, 13 Aug 2013 13:04:30 GMT. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH. Supported: replaces, timer. Content-Type: application/sdp. Content-Length: 264. . v=0. o=root 1263161426 1263161426 IN IP4 94.94.X.X. s=Asterisk PBX 11.3.0. c=IN IP4 94.94.X.X. t=0 0. m=audio 10782 RTP/AVP 0 110 101. a=rtpmap:0 PCMU/8000. a=rtpmap:110 speex/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=sendrecv.
# U 178.79.x.x:5060 -> 94.94.X.X:1025 SIP/2.0 407 Proxy Authentication Required. Via: SIP/2.0/UDP 94.94.X.X:5060;branch=z9hG4bK4a420cee;rport=1025. From: "asterisk" sip:512@test.domain;tag=as76007db0. To: sip:526@test.domain:5060;tag=333a0370df4a40d5d5a0c21bb156e2a6.4a00. Call-ID: 068a5a23639785a7583d952d6f9bca84@test.domain. CSeq: 102 INVITE. Proxy-Authenticate: Digest realm="test.domain", nonce="UgowYFIKLzQXDSUWUpy4xlhdXZ7WNjPt". Server: kamailio (4.0.2 (x86_64/linux)). Content-Length: 0. .
# U 94.94.X.X:1025 -> 178.79.x.x:5060 INVITE sip:526@test.domain:5060 SIP/2.0. Via: SIP/2.0/UDP 94.94.X.X:5060;branch=z9hG4bK4a420cee;rport. Max-Forwards: 70. From: "asterisk" sip:512@test.domain;tag=as76007db0. To: sip:526@test.domain:5060. Contact: sip:512@94.94.X.X:5060. Call-ID: 068a5a23639785a7583d952d6f9bca84@test.domain. CSeq: 102 INVITE. User-Agent: DICE 1.8.20.1. Date: Tue, 13 Aug 2013 13:04:30 GMT. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH. Supported: replaces, timer. Content-Type: application/sdp. Content-Length: 264. . v=0. o=root 1263161426 1263161426 IN IP4 94.94.X.X. s=Asterisk PBX 11.3.0. c=IN IP4 94.94.X.X. t=0 0. m=audio 10782 RTP/AVP 0 110 101. a=rtpmap:0 PCMU/8000. a=rtpmap:110 speex/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=sendrecv.
# U 178.79.x.x:5060 -> 94.94.X.X:1025 SIP/2.0 407 Proxy Authentication Required. Via: SIP/2.0/UDP 94.94.X.X:5060;branch=z9hG4bK4a420cee;rport=1025. From: "asterisk" sip:512@test.domain;tag=as76007db0. To: sip:526@test.domain:5060;tag=333a0370df4a40d5d5a0c21bb156e2a6.4a00. Call-ID: 068a5a23639785a7583d952d6f9bca84@test.domain. CSeq: 102 INVITE. Proxy-Authenticate: Digest realm="test.domain", nonce="UgowYFIKLzQXDSUWUpy4xlhdXZ7WNjPt". Server: kamailio (4.0.2 (x86_64/linux)). Content-Length: 0. .
# U 94.94.X.X:1025 -> 178.79.x.x:5060 ACK sip:526@test.domain:5060 SIP/2.0. Via: SIP/2.0/UDP 94.94.X.X:5060;branch=z9hG4bK4a420cee;rport. Max-Forwards: 70. From: "asterisk" sip:512@test.domain;tag=as76007db0. To: sip:526@test.domain:5060;tag=333a0370df4a40d5d5a0c21bb156e2a6.4a00. Contact: sip:512@94.94.X.X:5060. Call-ID: 068a5a23639785a7583d952d6f9bca84@test.domain. CSeq: 102 ACK. User-Agent: DICE 1.8.20.1. Content-Length: 0. .
# U 94.94.X.X:1025 -> 178.79.x.x:5060 ACK sip:526@test.domain:5060 SIP/2.0. Via: SIP/2.0/UDP 94.94.X.X:5060;branch=z9hG4bK4a420cee;rport. Max-Forwards: 70. From: "asterisk" sip:512@test.domain;tag=as76007db0. To: sip:526@test.domain:5060. Contact: sip:512@94.94.X.X:5060. Call-ID: 068a5a23639785a7583d952d6f9bca84@test.domain. CSeq: 102 ACK. User-Agent: DICE 1.8.20.1. Content-Length: 0. .
# T 178.79.x.x:5060 -> 94.94.X.X:61922 [AP] ACK sip:526@94.94.X.X:1238;transport=TCP;ob SIP/2.0. Via: SIP/2.0/TCP 178.79.x.x;branch=z9hG4bKcydzigwkX. Via: SIP/2.0/UDP 94.94.X.X:5060;branch=z9hG4bK4a420cee;rport=1025. Max-Forwards: 16. From: "asterisk" sip:512@test.domain;tag=as76007db0. To: sip:526@test.domain:5060. Contact: sip:512@94.94.X.X:1025. Call-ID: 068a5a23639785a7583d952d6f9bca84@test.domain. CSeq: 102 ACK. User-Agent: DICE 1.8.20.1. Content-Length: 0. .
# T 178.79.x.x:5060 -> 94.94.X.X:61922 [AP] ACK sip:526@94.94.X.X:1238;transport=TCP;ob SIP/2.0. Via: SIP/2.0/TCP 178.79.x.x;branch=z9hG4bKcydzigwkX. Via: SIP/2.0/UDP 94.94.X.X:5060;branch=z9hG4bK4a420cee;rport=1025. Max-Forwards: 16. From: "asterisk" sip:512@test.domain;tag=as76007db0. To: sip:526@test.domain:5060. Contact: sip:512@94.94.X.X:1025. Call-ID: 068a5a23639785a7583d952d6f9bca84@test.domain. CSeq: 102 ACK. User-Agent: DICE 1.8.20.1. Content-Length: 0. .
Cheers, Daniel
Cheers, Roberto Fichera.
Cheers, Daniel
On 7/30/13 6:44 PM, Roberto Fichera wrote:
Hi All,
Sorry for cross-posting this email to PJLIB, but maybe there are some things related. Anyhow! I'm having problems on kamailio v4.0.2 under Fedora 18 64bit and TCP client like iPhone using PJSIP as SIP library. Basically once the iPhone side in close the call (TCP->UDP) I'm getting the error below. Kamailio is running under a VPS without NATed network so it uses a real public address. Furthermore, note that tcp_main is answering to a 192.168.2.98 ip address which is the iPhone client. This looks really strange to me since it should answer directly to the public/port used for the registration and not to a such kind of reserved address. The kamilio configuration is basically the default with a very few changes like NAT, rtpproxy and postgresql backend.
This problems doesn't happen at all when using UDP->UDP calls. But I cannot use it because as you certain know UDP connection under iPhone will not work when the application run in background mode.
Can someone suggest how to solve this issue or maybe suggest a TCP working solution for iPhone?
Thanks in advance. Roberto Fichera.
Jul 30 16:21:53 proxy /usr/sbin/kamailio[9502]: ERROR: <core> [tcp_main.c:4432]: tcpconn_main_timeout(): connect 192.168.2.98:5060 failed (timeout) Jul 30 16:21:55 proxy rtpproxy[2262]: DBUG:get_command: received command "9483_9 D 12d1d19926c4ff742a52f0c855b1bb83@94.94.x.x:5060 as74e0c388 GROahimCK6KTrl5CkYEg7nuoPIIXZ8cj" Jul 30 16:21:55 proxy rtpproxy[2262]: INFO:handle_delete: forcefully deleting session 1 on ports 15604/17354 Jul 30 16:21:55 proxy rtpproxy[2262]: INFO:remove_session: RTP stats: 354 in from callee, 603 in from caller, 957 relayed, 0 dropped Jul 30 16:21:55 proxy rtpproxy[2262]: INFO:remove_session: RTCP stats: 5 in from callee, 2 in from caller, 7 relayed, 0 dropped Jul 30 16:21:55 proxy rtpproxy[2262]: INFO:remove_session: session on ports 15604/17354 is cleaned up Jul 30 16:21:55 proxy rtpproxy[2262]: DBUG:doreply: sending reply "9483_9 0 Jul 30 16:21:55 proxy rtpproxy[2262]: " Jul 30 16:22:04 proxy /usr/sbin/kamailio[9502]: ERROR: <core> [tcp_main.c:4432]: tcpconn_main_timeout(): connect 192.168.2.98:5060 failed (timeout) Jul 30 16:22:14 proxy rtpproxy[2262]: DBUG:get_command: received command "9496_16 D 12d1d19926c4ff742a52f0c855b1bb83@94.94.x.x:5060 GROahimCK6KTrl5CkYEg7nuoPIIXZ8cj as74e0c388" Jul 30 16:22:14 proxy rtpproxy[2262]: INFO:handle_command: delete request failed: session 12d1d19926c4ff742a52f0c855b1bb83@94.94.x.x:5060, tags GROahimCK6KTrl5CkYEg7nuoPIIXZ8cj/as74e0c388 not found Jul 30 16:22:14 proxy rtpproxy[2262]: DBUG:doreply: sending reply "9496_16 E8 Jul 30 16:22:14 proxy rtpproxy[2262]: "
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
On 08/13/2013 03:15 PM, Roberto Fichera wrote:
On 08/13/2013 02:33 PM, Daniel-Constantin Mierla wrote:
Hello,
On 8/13/13 1:10 PM, Roberto Fichera wrote:
On 08/13/2013 12:03 PM, Daniel-Constantin Mierla wrote:
Hello,
you should grab the ngrep for such call to understand better what happens. Also, dumping the location records will be useful (kamctl ul show).
Also, be sure that tcp connection lifetime is long enough to survive re-registration. To avoid trying to open connections behind nat, use set_forward_no_connect() for calls involving nat traversal.
I'm using the default conf coming from fedora rpm. So, mainly the problem seems related to kamailio which doesn't reuse the TCP port used by NATed clients. I've also notice that the received field isn't set at all, so this means that the contact will not get aliased at all.
I would really like to have a look to a working cfg file for TCP NATed clients that reuse the TCP port. Even better if the configuration is based on the fedora default rpm.
if received is not set, then means the register was not detected as coming from behind nat. Is the phone using stun?
I'm testing with a normal rtpproxy configuration. BTW udp -> udp work perfectly.
Again, put here the ngrep for registration and a call to see if something is wrong with signaling. There is no help that we can provide otherwise. The default config works fine for tcp and natted clients, I use it everywhere for this case without issues.
I tried the default cfg enabling both NAT and RTPproxy, but seems that kamailio doesn't reuse TCP ports. Anyway, this is a call from UDP (512) -> TCP (526) both behind the same NAT, from kamailio point of view
I forgot to say that the received field is now present because I've changed the route[NATDETECT] in the default configuration as
route[NATDETECT] { #!ifdef WITH_NAT force_rport();
-->>> if (nat_uac_test("19") || proto != UDP) { if (is_method("REGISTER")) { fix_nated_register(); } else { fix_nated_contact(); } setflag(FLT_NATS); } #!endif return; }
[root@proxy ~]# kamctl ul show 526 Contact:: sip:526@94.94.X.X:1238;transport=TCP;ob;q=;expires=537;flags=0x0;cflags=0x40;state=0;socket=tcp:178.79.x.x:5060;methods=0x1FDF;received=sip:94.94.X.X:61922;transport=TCP;user_agent=<DICE Smartphone 1.0/iPhone>;reg-id=0 [root@proxy ~]# kamctl ul show 512 Contact:: sip:512@94.94.X.X:5060;q=;expires=32;flags=0x0;cflags=0x40;state=0;socket=udp:178.79.x.x:5060;methods=0xFFFFFFFF;received=sip:94.94.X.X:1025;user_agent=<DICE 1.8.20.1>;reg-id=0 [root@proxy ~]#
U 94.94.X.X:1025 -> 178.79.x.x:5060 INVITE sip:526@test.domain:5060 SIP/2.0. Via: SIP/2.0/UDP 94.94.X.X:5060;branch=z9hG4bK4a420cee;rport. Max-Forwards: 70. From: "asterisk" sip:512@test.domain;tag=as76007db0. To: sip:526@test.domain:5060. Contact: sip:512@94.94.X.X:5060. Call-ID: 068a5a23639785a7583d952d6f9bca84@test.domain. CSeq: 102 INVITE. User-Agent: DICE 1.8.20.1. Date: Tue, 13 Aug 2013 13:04:30 GMT. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH. Supported: replaces, timer. Content-Type: application/sdp. Content-Length: 264. . v=0. o=root 1263161426 1263161426 IN IP4 94.94.X.X. s=Asterisk PBX 11.3.0. c=IN IP4 94.94.X.X. t=0 0. m=audio 10782 RTP/AVP 0 110 101. a=rtpmap:0 PCMU/8000. a=rtpmap:110 speex/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=sendrecv.
# U 178.79.x.x:5060 -> 94.94.X.X:1025 SIP/2.0 407 Proxy Authentication Required. Via: SIP/2.0/UDP 94.94.X.X:5060;branch=z9hG4bK4a420cee;rport=1025. From: "asterisk" sip:512@test.domain;tag=as76007db0. To: sip:526@test.domain:5060;tag=333a0370df4a40d5d5a0c21bb156e2a6.4a00. Call-ID: 068a5a23639785a7583d952d6f9bca84@test.domain. CSeq: 102 INVITE. Proxy-Authenticate: Digest realm="test.domain", nonce="UgowYFIKLzQXDSUWUpy4xlhdXZ7WNjPt". Server: kamailio (4.0.2 (x86_64/linux)). Content-Length: 0. .
# U 94.94.X.X:1025 -> 178.79.x.x:5060 INVITE sip:526@test.domain:5060 SIP/2.0. Via: SIP/2.0/UDP 94.94.X.X:5060;branch=z9hG4bK4a420cee;rport. Max-Forwards: 70. From: "asterisk" sip:512@test.domain;tag=as76007db0. To: sip:526@test.domain:5060. Contact: sip:512@94.94.X.X:5060. Call-ID: 068a5a23639785a7583d952d6f9bca84@test.domain. CSeq: 102 INVITE. User-Agent: DICE 1.8.20.1. Date: Tue, 13 Aug 2013 13:04:30 GMT. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH. Supported: replaces, timer. Content-Type: application/sdp. Content-Length: 264. . v=0. o=root 1263161426 1263161426 IN IP4 94.94.X.X. s=Asterisk PBX 11.3.0. c=IN IP4 94.94.X.X. t=0 0. m=audio 10782 RTP/AVP 0 110 101. a=rtpmap:0 PCMU/8000. a=rtpmap:110 speex/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=sendrecv.
# U 178.79.x.x:5060 -> 94.94.X.X:1025 SIP/2.0 407 Proxy Authentication Required. Via: SIP/2.0/UDP 94.94.X.X:5060;branch=z9hG4bK4a420cee;rport=1025. From: "asterisk" sip:512@test.domain;tag=as76007db0. To: sip:526@test.domain:5060;tag=333a0370df4a40d5d5a0c21bb156e2a6.4a00. Call-ID: 068a5a23639785a7583d952d6f9bca84@test.domain. CSeq: 102 INVITE. Proxy-Authenticate: Digest realm="test.domain", nonce="UgowYFIKLzQXDSUWUpy4xlhdXZ7WNjPt". Server: kamailio (4.0.2 (x86_64/linux)). Content-Length: 0. .
# U 94.94.X.X:1025 -> 178.79.x.x:5060 ACK sip:526@test.domain:5060 SIP/2.0. Via: SIP/2.0/UDP 94.94.X.X:5060;branch=z9hG4bK4a420cee;rport. Max-Forwards: 70. From: "asterisk" sip:512@test.domain;tag=as76007db0. To: sip:526@test.domain:5060;tag=333a0370df4a40d5d5a0c21bb156e2a6.4a00. Contact: sip:512@94.94.X.X:5060. Call-ID: 068a5a23639785a7583d952d6f9bca84@test.domain. CSeq: 102 ACK. User-Agent: DICE 1.8.20.1. Content-Length: 0. .
# U 94.94.X.X:1025 -> 178.79.x.x:5060 ACK sip:526@test.domain:5060 SIP/2.0. Via: SIP/2.0/UDP 94.94.X.X:5060;branch=z9hG4bK4a420cee;rport. Max-Forwards: 70. From: "asterisk" sip:512@test.domain;tag=as76007db0. To: sip:526@test.domain:5060. Contact: sip:512@94.94.X.X:5060. Call-ID: 068a5a23639785a7583d952d6f9bca84@test.domain. CSeq: 102 ACK. User-Agent: DICE 1.8.20.1. Content-Length: 0. .
# T 178.79.x.x:5060 -> 94.94.X.X:61922 [AP] ACK sip:526@94.94.X.X:1238;transport=TCP;ob SIP/2.0. Via: SIP/2.0/TCP 178.79.x.x;branch=z9hG4bKcydzigwkX. Via: SIP/2.0/UDP 94.94.X.X:5060;branch=z9hG4bK4a420cee;rport=1025. Max-Forwards: 16. From: "asterisk" sip:512@test.domain;tag=as76007db0. To: sip:526@test.domain:5060. Contact: sip:512@94.94.X.X:1025. Call-ID: 068a5a23639785a7583d952d6f9bca84@test.domain. CSeq: 102 ACK. User-Agent: DICE 1.8.20.1. Content-Length: 0. .
# T 178.79.x.x:5060 -> 94.94.X.X:61922 [AP] ACK sip:526@94.94.X.X:1238;transport=TCP;ob SIP/2.0. Via: SIP/2.0/TCP 178.79.x.x;branch=z9hG4bKcydzigwkX. Via: SIP/2.0/UDP 94.94.X.X:5060;branch=z9hG4bK4a420cee;rport=1025. Max-Forwards: 16. From: "asterisk" sip:512@test.domain;tag=as76007db0. To: sip:526@test.domain:5060. Contact: sip:512@94.94.X.X:1025. Call-ID: 068a5a23639785a7583d952d6f9bca84@test.domain. CSeq: 102 ACK. User-Agent: DICE 1.8.20.1. Content-Length: 0. .
Cheers, Daniel
Cheers, Roberto Fichera.
Cheers, Daniel
On 7/30/13 6:44 PM, Roberto Fichera wrote:
Hi All,
Sorry for cross-posting this email to PJLIB, but maybe there are some things related. Anyhow! I'm having problems on kamailio v4.0.2 under Fedora 18 64bit and TCP client like iPhone using PJSIP as SIP library. Basically once the iPhone side in close the call (TCP->UDP) I'm getting the error below. Kamailio is running under a VPS without NATed network so it uses a real public address. Furthermore, note that tcp_main is answering to a 192.168.2.98 ip address which is the iPhone client. This looks really strange to me since it should answer directly to the public/port used for the registration and not to a such kind of reserved address. The kamilio configuration is basically the default with a very few changes like NAT, rtpproxy and postgresql backend.
This problems doesn't happen at all when using UDP->UDP calls. But I cannot use it because as you certain know UDP connection under iPhone will not work when the application run in background mode.
Can someone suggest how to solve this issue or maybe suggest a TCP working solution for iPhone?
Thanks in advance. Roberto Fichera.
Jul 30 16:21:53 proxy /usr/sbin/kamailio[9502]: ERROR: <core> [tcp_main.c:4432]: tcpconn_main_timeout(): connect 192.168.2.98:5060 failed (timeout) Jul 30 16:21:55 proxy rtpproxy[2262]: DBUG:get_command: received command "9483_9 D 12d1d19926c4ff742a52f0c855b1bb83@94.94.x.x:5060 as74e0c388 GROahimCK6KTrl5CkYEg7nuoPIIXZ8cj" Jul 30 16:21:55 proxy rtpproxy[2262]: INFO:handle_delete: forcefully deleting session 1 on ports 15604/17354 Jul 30 16:21:55 proxy rtpproxy[2262]: INFO:remove_session: RTP stats: 354 in from callee, 603 in from caller, 957 relayed, 0 dropped Jul 30 16:21:55 proxy rtpproxy[2262]: INFO:remove_session: RTCP stats: 5 in from callee, 2 in from caller, 7 relayed, 0 dropped Jul 30 16:21:55 proxy rtpproxy[2262]: INFO:remove_session: session on ports 15604/17354 is cleaned up Jul 30 16:21:55 proxy rtpproxy[2262]: DBUG:doreply: sending reply "9483_9 0 Jul 30 16:21:55 proxy rtpproxy[2262]: " Jul 30 16:22:04 proxy /usr/sbin/kamailio[9502]: ERROR: <core> [tcp_main.c:4432]: tcpconn_main_timeout(): connect 192.168.2.98:5060 failed (timeout) Jul 30 16:22:14 proxy rtpproxy[2262]: DBUG:get_command: received command "9496_16 D 12d1d19926c4ff742a52f0c855b1bb83@94.94.x.x:5060 GROahimCK6KTrl5CkYEg7nuoPIIXZ8cj as74e0c388" Jul 30 16:22:14 proxy rtpproxy[2262]: INFO:handle_command: delete request failed: session 12d1d19926c4ff742a52f0c855b1bb83@94.94.x.x:5060, tags GROahimCK6KTrl5CkYEg7nuoPIIXZ8cj/as74e0c388 not found Jul 30 16:22:14 proxy rtpproxy[2262]: DBUG:doreply: sending reply "9496_16 E8 Jul 30 16:22:14 proxy rtpproxy[2262]: "
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Can you get a ngrep trace for a registration as well (for the phone using tcp)?
Daniel
On 8/13/13 3:23 PM, Roberto Fichera wrote:
On 08/13/2013 03:15 PM, Roberto Fichera wrote:
On 08/13/2013 02:33 PM, Daniel-Constantin Mierla wrote:
Hello,
On 8/13/13 1:10 PM, Roberto Fichera wrote:
On 08/13/2013 12:03 PM, Daniel-Constantin Mierla wrote:
Hello,
you should grab the ngrep for such call to understand better what happens. Also, dumping the location records will be useful (kamctl ul show).
Also, be sure that tcp connection lifetime is long enough to survive re-registration. To avoid trying to open connections behind nat, use set_forward_no_connect() for calls involving nat traversal.
I'm using the default conf coming from fedora rpm. So, mainly the problem seems related to kamailio which doesn't reuse the TCP port used by NATed clients. I've also notice that the received field isn't set at all, so this means that the contact will not get aliased at all.
I would really like to have a look to a working cfg file for TCP NATed clients that reuse the TCP port. Even better if the configuration is based on the fedora default rpm.
if received is not set, then means the register was not detected as coming from behind nat. Is the phone using stun?
I'm testing with a normal rtpproxy configuration. BTW udp -> udp work perfectly.
Again, put here the ngrep for registration and a call to see if something is wrong with signaling. There is no help that we can provide otherwise. The default config works fine for tcp and natted clients, I use it everywhere for this case without issues.
I tried the default cfg enabling both NAT and RTPproxy, but seems that kamailio doesn't reuse TCP ports. Anyway, this is a call from UDP (512) -> TCP (526) both behind the same NAT, from kamailio point of view
I forgot to say that the received field is now present because I've changed the route[NATDETECT] in the default configuration as
route[NATDETECT] { #!ifdef WITH_NAT force_rport();
-->>> if (nat_uac_test("19") || proto != UDP) { if (is_method("REGISTER")) { fix_nated_register(); } else { fix_nated_contact(); } setflag(FLT_NATS); } #!endif return; }
[root@proxy ~]# kamctl ul show 526 Contact:: sip:526@94.94.X.X:1238;transport=TCP;ob;q=;expires=537;flags=0x0;cflags=0x40;state=0;socket=tcp:178.79.x.x:5060;methods=0x1FDF;received=sip:94.94.X.X:61922;transport=TCP;user_agent=<DICE Smartphone 1.0/iPhone>;reg-id=0 [root@proxy ~]# kamctl ul show 512 Contact:: sip:512@94.94.X.X:5060;q=;expires=32;flags=0x0;cflags=0x40;state=0;socket=udp:178.79.x.x:5060;methods=0xFFFFFFFF;received=sip:94.94.X.X:1025;user_agent=<DICE 1.8.20.1>;reg-id=0 [root@proxy ~]#
U 94.94.X.X:1025 -> 178.79.x.x:5060 INVITE sip:526@test.domain:5060 SIP/2.0. Via: SIP/2.0/UDP 94.94.X.X:5060;branch=z9hG4bK4a420cee;rport. Max-Forwards: 70. From: "asterisk" sip:512@test.domain;tag=as76007db0. To: sip:526@test.domain:5060. Contact: sip:512@94.94.X.X:5060. Call-ID: 068a5a23639785a7583d952d6f9bca84@test.domain. CSeq: 102 INVITE. User-Agent: DICE 1.8.20.1. Date: Tue, 13 Aug 2013 13:04:30 GMT. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH. Supported: replaces, timer. Content-Type: application/sdp. Content-Length: 264. . v=0. o=root 1263161426 1263161426 IN IP4 94.94.X.X. s=Asterisk PBX 11.3.0. c=IN IP4 94.94.X.X. t=0 0. m=audio 10782 RTP/AVP 0 110 101. a=rtpmap:0 PCMU/8000. a=rtpmap:110 speex/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=sendrecv.
# U 178.79.x.x:5060 -> 94.94.X.X:1025 SIP/2.0 407 Proxy Authentication Required. Via: SIP/2.0/UDP 94.94.X.X:5060;branch=z9hG4bK4a420cee;rport=1025. From: "asterisk" sip:512@test.domain;tag=as76007db0. To: sip:526@test.domain:5060;tag=333a0370df4a40d5d5a0c21bb156e2a6.4a00. Call-ID: 068a5a23639785a7583d952d6f9bca84@test.domain. CSeq: 102 INVITE. Proxy-Authenticate: Digest realm="test.domain", nonce="UgowYFIKLzQXDSUWUpy4xlhdXZ7WNjPt". Server: kamailio (4.0.2 (x86_64/linux)). Content-Length: 0. .
# U 94.94.X.X:1025 -> 178.79.x.x:5060 INVITE sip:526@test.domain:5060 SIP/2.0. Via: SIP/2.0/UDP 94.94.X.X:5060;branch=z9hG4bK4a420cee;rport. Max-Forwards: 70. From: "asterisk" sip:512@test.domain;tag=as76007db0. To: sip:526@test.domain:5060. Contact: sip:512@94.94.X.X:5060. Call-ID: 068a5a23639785a7583d952d6f9bca84@test.domain. CSeq: 102 INVITE. User-Agent: DICE 1.8.20.1. Date: Tue, 13 Aug 2013 13:04:30 GMT. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH. Supported: replaces, timer. Content-Type: application/sdp. Content-Length: 264. . v=0. o=root 1263161426 1263161426 IN IP4 94.94.X.X. s=Asterisk PBX 11.3.0. c=IN IP4 94.94.X.X. t=0 0. m=audio 10782 RTP/AVP 0 110 101. a=rtpmap:0 PCMU/8000. a=rtpmap:110 speex/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=sendrecv.
# U 178.79.x.x:5060 -> 94.94.X.X:1025 SIP/2.0 407 Proxy Authentication Required. Via: SIP/2.0/UDP 94.94.X.X:5060;branch=z9hG4bK4a420cee;rport=1025. From: "asterisk" sip:512@test.domain;tag=as76007db0. To: sip:526@test.domain:5060;tag=333a0370df4a40d5d5a0c21bb156e2a6.4a00. Call-ID: 068a5a23639785a7583d952d6f9bca84@test.domain. CSeq: 102 INVITE. Proxy-Authenticate: Digest realm="test.domain", nonce="UgowYFIKLzQXDSUWUpy4xlhdXZ7WNjPt". Server: kamailio (4.0.2 (x86_64/linux)). Content-Length: 0. .
# U 94.94.X.X:1025 -> 178.79.x.x:5060 ACK sip:526@test.domain:5060 SIP/2.0. Via: SIP/2.0/UDP 94.94.X.X:5060;branch=z9hG4bK4a420cee;rport. Max-Forwards: 70. From: "asterisk" sip:512@test.domain;tag=as76007db0. To: sip:526@test.domain:5060;tag=333a0370df4a40d5d5a0c21bb156e2a6.4a00. Contact: sip:512@94.94.X.X:5060. Call-ID: 068a5a23639785a7583d952d6f9bca84@test.domain. CSeq: 102 ACK. User-Agent: DICE 1.8.20.1. Content-Length: 0. .
# U 94.94.X.X:1025 -> 178.79.x.x:5060 ACK sip:526@test.domain:5060 SIP/2.0. Via: SIP/2.0/UDP 94.94.X.X:5060;branch=z9hG4bK4a420cee;rport. Max-Forwards: 70. From: "asterisk" sip:512@test.domain;tag=as76007db0. To: sip:526@test.domain:5060. Contact: sip:512@94.94.X.X:5060. Call-ID: 068a5a23639785a7583d952d6f9bca84@test.domain. CSeq: 102 ACK. User-Agent: DICE 1.8.20.1. Content-Length: 0. .
# T 178.79.x.x:5060 -> 94.94.X.X:61922 [AP] ACK sip:526@94.94.X.X:1238;transport=TCP;ob SIP/2.0. Via: SIP/2.0/TCP 178.79.x.x;branch=z9hG4bKcydzigwkX. Via: SIP/2.0/UDP 94.94.X.X:5060;branch=z9hG4bK4a420cee;rport=1025. Max-Forwards: 16. From: "asterisk" sip:512@test.domain;tag=as76007db0. To: sip:526@test.domain:5060. Contact: sip:512@94.94.X.X:1025. Call-ID: 068a5a23639785a7583d952d6f9bca84@test.domain. CSeq: 102 ACK. User-Agent: DICE 1.8.20.1. Content-Length: 0. .
# T 178.79.x.x:5060 -> 94.94.X.X:61922 [AP] ACK sip:526@94.94.X.X:1238;transport=TCP;ob SIP/2.0. Via: SIP/2.0/TCP 178.79.x.x;branch=z9hG4bKcydzigwkX. Via: SIP/2.0/UDP 94.94.X.X:5060;branch=z9hG4bK4a420cee;rport=1025. Max-Forwards: 16. From: "asterisk" sip:512@test.domain;tag=as76007db0. To: sip:526@test.domain:5060. Contact: sip:512@94.94.X.X:1025. Call-ID: 068a5a23639785a7583d952d6f9bca84@test.domain. CSeq: 102 ACK. User-Agent: DICE 1.8.20.1. Content-Length: 0. .
Cheers, Daniel
Cheers, Roberto Fichera.
Cheers, Daniel
On 7/30/13 6:44 PM, Roberto Fichera wrote:
Hi All,
Sorry for cross-posting this email to PJLIB, but maybe there are some things related. Anyhow! I'm having problems on kamailio v4.0.2 under Fedora 18 64bit and TCP client like iPhone using PJSIP as SIP library. Basically once the iPhone side in close the call (TCP->UDP) I'm getting the error below. Kamailio is running under a VPS without NATed network so it uses a real public address. Furthermore, note that tcp_main is answering to a 192.168.2.98 ip address which is the iPhone client. This looks really strange to me since it should answer directly to the public/port used for the registration and not to a such kind of reserved address. The kamilio configuration is basically the default with a very few changes like NAT, rtpproxy and postgresql backend.
This problems doesn't happen at all when using UDP->UDP calls. But I cannot use it because as you certain know UDP connection under iPhone will not work when the application run in background mode.
Can someone suggest how to solve this issue or maybe suggest a TCP working solution for iPhone?
Thanks in advance. Roberto Fichera.
Jul 30 16:21:53 proxy /usr/sbin/kamailio[9502]: ERROR: <core> [tcp_main.c:4432]: tcpconn_main_timeout(): connect 192.168.2.98:5060 failed (timeout) Jul 30 16:21:55 proxy rtpproxy[2262]: DBUG:get_command: received command "9483_9 D 12d1d19926c4ff742a52f0c855b1bb83@94.94.x.x:5060 as74e0c388 GROahimCK6KTrl5CkYEg7nuoPIIXZ8cj" Jul 30 16:21:55 proxy rtpproxy[2262]: INFO:handle_delete: forcefully deleting session 1 on ports 15604/17354 Jul 30 16:21:55 proxy rtpproxy[2262]: INFO:remove_session: RTP stats: 354 in from callee, 603 in from caller, 957 relayed, 0 dropped Jul 30 16:21:55 proxy rtpproxy[2262]: INFO:remove_session: RTCP stats: 5 in from callee, 2 in from caller, 7 relayed, 0 dropped Jul 30 16:21:55 proxy rtpproxy[2262]: INFO:remove_session: session on ports 15604/17354 is cleaned up Jul 30 16:21:55 proxy rtpproxy[2262]: DBUG:doreply: sending reply "9483_9 0 Jul 30 16:21:55 proxy rtpproxy[2262]: " Jul 30 16:22:04 proxy /usr/sbin/kamailio[9502]: ERROR: <core> [tcp_main.c:4432]: tcpconn_main_timeout(): connect 192.168.2.98:5060 failed (timeout) Jul 30 16:22:14 proxy rtpproxy[2262]: DBUG:get_command: received command "9496_16 D 12d1d19926c4ff742a52f0c855b1bb83@94.94.x.x:5060 GROahimCK6KTrl5CkYEg7nuoPIIXZ8cj as74e0c388" Jul 30 16:22:14 proxy rtpproxy[2262]: INFO:handle_command: delete request failed: session 12d1d19926c4ff742a52f0c855b1bb83@94.94.x.x:5060, tags GROahimCK6KTrl5CkYEg7nuoPIIXZ8cj/as74e0c388 not found Jul 30 16:22:14 proxy rtpproxy[2262]: DBUG:doreply: sending reply "9496_16 E8 Jul 30 16:22:14 proxy rtpproxy[2262]: "
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
On 08/13/2013 03:25 PM, Daniel-Constantin Mierla wrote:
Can you get a ngrep trace for a registration as well (for the phone using tcp)?
Ok! I'll use pjsua from my local machine connecting in the same way as the TCP client was doing. The TCP client it's an iPhone using the same pjlib library.
Daniel
On 8/13/13 3:23 PM, Roberto Fichera wrote:
On 08/13/2013 03:15 PM, Roberto Fichera wrote:
On 08/13/2013 02:33 PM, Daniel-Constantin Mierla wrote:
Hello,
On 8/13/13 1:10 PM, Roberto Fichera wrote:
On 08/13/2013 12:03 PM, Daniel-Constantin Mierla wrote:
Hello,
you should grab the ngrep for such call to understand better what happens. Also, dumping the location records will be useful (kamctl ul show).
Also, be sure that tcp connection lifetime is long enough to survive re-registration. To avoid trying to open connections behind nat, use set_forward_no_connect() for calls involving nat traversal.
I'm using the default conf coming from fedora rpm. So, mainly the problem seems related to kamailio which doesn't reuse the TCP port used by NATed clients. I've also notice that the received field isn't set at all, so this means that the contact will not get aliased at all.
I would really like to have a look to a working cfg file for TCP NATed clients that reuse the TCP port. Even better if the configuration is based on the fedora default rpm.
if received is not set, then means the register was not detected as coming from behind nat. Is the phone using stun?
I'm testing with a normal rtpproxy configuration. BTW udp -> udp work perfectly.
Again, put here the ngrep for registration and a call to see if something is wrong with signaling. There is no help that we can provide otherwise. The default config works fine for tcp and natted clients, I use it everywhere for this case without issues.
I tried the default cfg enabling both NAT and RTPproxy, but seems that kamailio doesn't reuse TCP ports. Anyway, this is a call from UDP (512) -> TCP (526) both behind the same NAT, from kamailio point of view
I forgot to say that the received field is now present because I've changed the route[NATDETECT] in the default configuration as
route[NATDETECT] { #!ifdef WITH_NAT force_rport();
-->>> if (nat_uac_test("19") || proto != UDP) { if (is_method("REGISTER")) { fix_nated_register(); } else { fix_nated_contact(); } setflag(FLT_NATS); } #!endif return; }
[root@proxy ~]# kamctl ul show 526 Contact:: sip:526@94.94.X.X:1238;transport=TCP;ob;q=;expires=537;flags=0x0;cflags=0x40;state=0;socket=tcp:178.79.x.x:5060;methods=0x1FDF;received=sip:94.94.X.X:61922;transport=TCP;user_agent=<DICE
Smartphone 1.0/iPhone>;reg-id=0 [root@proxy ~]# kamctl ul show 512 Contact:: sip:512@94.94.X.X:5060;q=;expires=32;flags=0x0;cflags=0x40;state=0;socket=udp:178.79.x.x:5060;methods=0xFFFFFFFF;received=sip:94.94.X.X:1025;user_agent=<DICE
1.8.20.1>;reg-id=0 [root@proxy ~]#
U 94.94.X.X:1025 -> 178.79.x.x:5060 INVITE sip:526@test.domain:5060 SIP/2.0. Via: SIP/2.0/UDP 94.94.X.X:5060;branch=z9hG4bK4a420cee;rport. Max-Forwards: 70. From: "asterisk" sip:512@test.domain;tag=as76007db0. To: sip:526@test.domain:5060. Contact: sip:512@94.94.X.X:5060. Call-ID: 068a5a23639785a7583d952d6f9bca84@test.domain. CSeq: 102 INVITE. User-Agent: DICE 1.8.20.1. Date: Tue, 13 Aug 2013 13:04:30 GMT. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH. Supported: replaces, timer. Content-Type: application/sdp. Content-Length: 264. . v=0. o=root 1263161426 1263161426 IN IP4 94.94.X.X. s=Asterisk PBX 11.3.0. c=IN IP4 94.94.X.X. t=0 0. m=audio 10782 RTP/AVP 0 110 101. a=rtpmap:0 PCMU/8000. a=rtpmap:110 speex/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=sendrecv.
# U 178.79.x.x:5060 -> 94.94.X.X:1025 SIP/2.0 407 Proxy Authentication Required. Via: SIP/2.0/UDP 94.94.X.X:5060;branch=z9hG4bK4a420cee;rport=1025. From: "asterisk" sip:512@test.domain;tag=as76007db0. To: sip:526@test.domain:5060;tag=333a0370df4a40d5d5a0c21bb156e2a6.4a00. Call-ID: 068a5a23639785a7583d952d6f9bca84@test.domain. CSeq: 102 INVITE. Proxy-Authenticate: Digest realm="test.domain", nonce="UgowYFIKLzQXDSUWUpy4xlhdXZ7WNjPt". Server: kamailio (4.0.2 (x86_64/linux)). Content-Length: 0. .
# U 94.94.X.X:1025 -> 178.79.x.x:5060 INVITE sip:526@test.domain:5060 SIP/2.0. Via: SIP/2.0/UDP 94.94.X.X:5060;branch=z9hG4bK4a420cee;rport. Max-Forwards: 70. From: "asterisk" sip:512@test.domain;tag=as76007db0. To: sip:526@test.domain:5060. Contact: sip:512@94.94.X.X:5060. Call-ID: 068a5a23639785a7583d952d6f9bca84@test.domain. CSeq: 102 INVITE. User-Agent: DICE 1.8.20.1. Date: Tue, 13 Aug 2013 13:04:30 GMT. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH. Supported: replaces, timer. Content-Type: application/sdp. Content-Length: 264. . v=0. o=root 1263161426 1263161426 IN IP4 94.94.X.X. s=Asterisk PBX 11.3.0. c=IN IP4 94.94.X.X. t=0 0. m=audio 10782 RTP/AVP 0 110 101. a=rtpmap:0 PCMU/8000. a=rtpmap:110 speex/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=sendrecv.
# U 178.79.x.x:5060 -> 94.94.X.X:1025 SIP/2.0 407 Proxy Authentication Required. Via: SIP/2.0/UDP 94.94.X.X:5060;branch=z9hG4bK4a420cee;rport=1025. From: "asterisk" sip:512@test.domain;tag=as76007db0. To: sip:526@test.domain:5060;tag=333a0370df4a40d5d5a0c21bb156e2a6.4a00. Call-ID: 068a5a23639785a7583d952d6f9bca84@test.domain. CSeq: 102 INVITE. Proxy-Authenticate: Digest realm="test.domain", nonce="UgowYFIKLzQXDSUWUpy4xlhdXZ7WNjPt". Server: kamailio (4.0.2 (x86_64/linux)). Content-Length: 0. .
# U 94.94.X.X:1025 -> 178.79.x.x:5060 ACK sip:526@test.domain:5060 SIP/2.0. Via: SIP/2.0/UDP 94.94.X.X:5060;branch=z9hG4bK4a420cee;rport. Max-Forwards: 70. From: "asterisk" sip:512@test.domain;tag=as76007db0. To: sip:526@test.domain:5060;tag=333a0370df4a40d5d5a0c21bb156e2a6.4a00. Contact: sip:512@94.94.X.X:5060. Call-ID: 068a5a23639785a7583d952d6f9bca84@test.domain. CSeq: 102 ACK. User-Agent: DICE 1.8.20.1. Content-Length: 0. .
# U 94.94.X.X:1025 -> 178.79.x.x:5060 ACK sip:526@test.domain:5060 SIP/2.0. Via: SIP/2.0/UDP 94.94.X.X:5060;branch=z9hG4bK4a420cee;rport. Max-Forwards: 70. From: "asterisk" sip:512@test.domain;tag=as76007db0. To: sip:526@test.domain:5060. Contact: sip:512@94.94.X.X:5060. Call-ID: 068a5a23639785a7583d952d6f9bca84@test.domain. CSeq: 102 ACK. User-Agent: DICE 1.8.20.1. Content-Length: 0. .
# T 178.79.x.x:5060 -> 94.94.X.X:61922 [AP] ACK sip:526@94.94.X.X:1238;transport=TCP;ob SIP/2.0. Via: SIP/2.0/TCP 178.79.x.x;branch=z9hG4bKcydzigwkX. Via: SIP/2.0/UDP 94.94.X.X:5060;branch=z9hG4bK4a420cee;rport=1025. Max-Forwards: 16. From: "asterisk" sip:512@test.domain;tag=as76007db0. To: sip:526@test.domain:5060. Contact: sip:512@94.94.X.X:1025. Call-ID: 068a5a23639785a7583d952d6f9bca84@test.domain. CSeq: 102 ACK. User-Agent: DICE 1.8.20.1. Content-Length: 0. .
# T 178.79.x.x:5060 -> 94.94.X.X:61922 [AP] ACK sip:526@94.94.X.X:1238;transport=TCP;ob SIP/2.0. Via: SIP/2.0/TCP 178.79.x.x;branch=z9hG4bKcydzigwkX. Via: SIP/2.0/UDP 94.94.X.X:5060;branch=z9hG4bK4a420cee;rport=1025. Max-Forwards: 16. From: "asterisk" sip:512@test.domain;tag=as76007db0. To: sip:526@test.domain:5060. Contact: sip:512@94.94.X.X:1025. Call-ID: 068a5a23639785a7583d952d6f9bca84@test.domain. CSeq: 102 ACK. User-Agent: DICE 1.8.20.1. Content-Length: 0. .
Cheers, Daniel
Cheers, Roberto Fichera.
Cheers, Daniel
On 7/30/13 6:44 PM, Roberto Fichera wrote: > Hi All, > > Sorry for cross-posting this email to PJLIB, but maybe there are some things related. > Anyhow! I'm having problems on kamailio v4.0.2 under Fedora 18 64bit and TCP client like iPhone using PJSIP as SIP > library. > Basically once the iPhone side in close the call (TCP->UDP) I'm getting the error below. Kamailio is running under > a VPS > without > NATed network so it uses a real public address. Furthermore, note that tcp_main is answering to a 192.168.2.98 ip > address > which is the iPhone client. This looks really strange to me since it should answer directly to the public/port used > for > the registration > and not to a such kind of reserved address. The kamilio configuration is basically the default with a very few > changes > like NAT, rtpproxy and postgresql backend. > > This problems doesn't happen at all when using UDP->UDP calls. But I cannot use it because as you certain know UDP > connection under iPhone will not work when the application run in background mode. > > Can someone suggest how to solve this issue or maybe suggest a TCP working solution for iPhone? > > Thanks in advance. > Roberto Fichera. > > Jul 30 16:21:53 proxy /usr/sbin/kamailio[9502]: ERROR: <core> [tcp_main.c:4432]: tcpconn_main_timeout(): connect > 192.168.2.98:5060 failed (timeout) > Jul 30 16:21:55 proxy rtpproxy[2262]: DBUG:get_command: received command "9483_9 D > 12d1d19926c4ff742a52f0c855b1bb83@94.94.x.x:5060 as74e0c388 GROahimCK6KTrl5CkYEg7nuoPIIXZ8cj" > Jul 30 16:21:55 proxy rtpproxy[2262]: INFO:handle_delete: forcefully deleting session 1 on ports 15604/17354 > Jul 30 16:21:55 proxy rtpproxy[2262]: INFO:remove_session: RTP stats: 354 in from callee, 603 in from caller, 957 > relayed, 0 dropped > Jul 30 16:21:55 proxy rtpproxy[2262]: INFO:remove_session: RTCP stats: 5 in from callee, 2 in from caller, 7 > relayed, 0 > dropped > Jul 30 16:21:55 proxy rtpproxy[2262]: INFO:remove_session: session on ports 15604/17354 is cleaned up > Jul 30 16:21:55 proxy rtpproxy[2262]: DBUG:doreply: sending reply "9483_9 0 > Jul 30 16:21:55 proxy rtpproxy[2262]: " > Jul 30 16:22:04 proxy /usr/sbin/kamailio[9502]: ERROR: <core> [tcp_main.c:4432]: tcpconn_main_timeout(): connect > 192.168.2.98:5060 failed (timeout) > Jul 30 16:22:14 proxy rtpproxy[2262]: DBUG:get_command: received command "9496_16 D > 12d1d19926c4ff742a52f0c855b1bb83@94.94.x.x:5060 GROahimCK6KTrl5CkYEg7nuoPIIXZ8cj as74e0c388" > Jul 30 16:22:14 proxy rtpproxy[2262]: INFO:handle_command: delete request failed: session > 12d1d19926c4ff742a52f0c855b1bb83@94.94.x.x:5060, tags GROahimCK6KTrl5CkYEg7nuoPIIXZ8cj/as74e0c388 not found > Jul 30 16:22:14 proxy rtpproxy[2262]: DBUG:doreply: sending reply "9496_16 E8 > Jul 30 16:22:14 proxy rtpproxy[2262]: " > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users@lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
On 08/13/2013 03:32 PM, Roberto Fichera wrote:
On 08/13/2013 03:25 PM, Daniel-Constantin Mierla wrote:
Can you get a ngrep trace for a registration as well (for the phone using tcp)?
Ok! I'll use pjsua from my local machine connecting in the same way as the TCP client was doing. The TCP client it's an iPhone using the same pjlib library.
I can confirm that the default cfg for TCP client doesn't work for me. My cfg file is attached. The TCP client doesn't receive any package at INVITE. Finally in /var/log/message I get this log below:
Aug 13 14:05:37 proxy /usr/sbin/kamailio[8401]: ERROR: <core> [tcp_main.c:4247]: handle_tcpconn_ev(): connect 94.94.X.X:1274 failed
Contact:: sip:528@94.94.X.X:1274;transport=TCP;ob;q=;expires=38;flags=0x0;cflags=0x0;state=0;socket=tcp:178.79.X.X:5060;methods=0x1FDF;user_agent=<PJSUA v2.1-svn Linux-3.10.5.201/x86_64/glibc-2.17>;reg-id=0 [root@proxy ~]# kamctl ul show 512 Contact:: sip:512@94.94.X.X:5060;q=;expires=61;flags=0x0;cflags=0x40;state=0;socket=udp:178.79.X.X:5060;methods=0xFFFFFFFF;received=sip:94.94.X.X:1025 user_agent=<DICE 1.8.20.1>;reg-id=0
[root@proxy ~]# ngrep -W byline -d eth0 port 5060 interface: eth0 (178.79.X.X/255.255.255.0) filter: (ip or ip6) and ( port 5060 ) # T 94.94.X.X:49519 -> 178.79.x.x:5060 [AP] REGISTER sip:test.domain;transport=tcp;hide SIP/2.0. Via: SIP/2.0/TCP 94.94.X.X:49519;rport;branch=z9hG4bKPje03c11fd-d742-4502-98d8-69ca456ddd56;alias. Max-Forwards: 70. From: sip:528@test.domain;hide;tag=64c9e9b9-cb23-4a79-bb0d-2497135e449b. To: sip:528@test.domain;hide. Call-ID: 3ebadaeb-e6d6-42e3-9362-5ca4b5ddd8e1. CSeq: 30034 REGISTER. User-Agent: PJSUA v2.1-svn Linux-3.10.5.201/x86_64/glibc-2.17. Contact: sip:528@94.94.X.X:1274;transport=TCP;ob. Expires: 300. Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS. Content-Length: 0. .
# T 178.79.x.x:5060 -> 94.94.X.X:49519 [AP] SIP/2.0 401 Unauthorized. Via: SIP/2.0/TCP 94.94.X.X:49519;rport=49519;branch=z9hG4bKPje03c11fd-d742-4502-98d8-69ca456ddd56;alias. From: sip:528@test.domain;hide;tag=64c9e9b9-cb23-4a79-bb0d-2497135e449b. To: sip:528@test.domain;hide;tag=333a0370df4a40d5d5a0c21bb156e2a6.9167. Call-ID: 3ebadaeb-e6d6-42e3-9362-5ca4b5ddd8e1. CSeq: 30034 REGISTER. WWW-Authenticate: Digest realm="test.domain", nonce="Ugo+VFIKPSgVy8fh9fSy9SDLvT0wO4QV". Server: kamailio (4.0.2 (x86_64/linux)). Content-Length: 0. .
# U 178.79.x.x:5060 -> 94.94.X.X:1025 .... ## T 94.94.X.X:49519 -> 178.79.x.x:5060 [AP] REGISTER sip:test.domain;transport=tcp;hide SIP/2.0. Via: SIP/2.0/TCP 94.94.X.X:49519;rport;branch=z9hG4bKPj68a9c487-ae8c-428a-8c50-dd195a46a6bf;alias. Max-Forwards: 70. From: sip:528@test.domain;hide;tag=64c9e9b9-cb23-4a79-bb0d-2497135e449b. To: sip:528@test.domain;hide. Call-ID: 3ebadaeb-e6d6-42e3-9362-5ca4b5ddd8e1. CSeq: 30035 REGISTER. User-Agent: PJSUA v2.1-svn Linux-3.10.5.201/x86_64/glibc-2.17. Contact: sip:528@94.94.X.X:1274;transport=TCP;ob. Expires: 300. Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS. Authorization: Digest username="528", realm="test.domain", nonce="Ugo+VFIKPSgVy8fh9fSy9SDLvT0wO4QV", uri="sip:test.domain;transport=tcp;hide", response="ac1c7311ccb887fc8fb494d8ebf1bd36". Content-Length: 0. .
# T 178.79.x.x:5060 -> 94.94.X.X:49519 [AP] SIP/2.0 200 OK. Via: SIP/2.0/TCP 94.94.X.X:49519;rport=49519;branch=z9hG4bKPj68a9c487-ae8c-428a-8c50-dd195a46a6bf;alias. From: sip:528@test.domain;hide;tag=64c9e9b9-cb23-4a79-bb0d-2497135e449b. To: sip:528@test.domain;hide;tag=333a0370df4a40d5d5a0c21bb156e2a6.5440. Call-ID: 3ebadaeb-e6d6-42e3-9362-5ca4b5ddd8e1. CSeq: 30035 REGISTER. Contact: sip:528@94.94.X.X:1274;transport=TCP;ob;expires=300. Server: kamailio (4.0.2 (x86_64/linux)). Content-Length: 0. .
## U 94.94.X.X:1025 -> 178.79.x.x:5060 INVITE sip:528@test.domain:5060 SIP/2.0. Via: SIP/2.0/UDP 94.94.X.X:5060;branch=z9hG4bK69275d13;rport. Max-Forwards: 70. From: "asterisk" sip:512@test.domain;tag=as4531a4ee. To: sip:528@test.domain:5060. Contact: sip:512@94.94.X.X:5060. Call-ID: 2ea7e0f3173949d40b69385b62b79d6a@test.domain. CSeq: 102 INVITE. User-Agent: DICE 1.8.20.1. Date: Tue, 13 Aug 2013 14:04:11 GMT. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH. Supported: replaces, timer. Content-Type: application/sdp. Content-Length: 262. . v=0. o=root 883039875 883039875 IN IP4 94.94.X.X. s=Asterisk PBX 11.3.0. c=IN IP4 94.94.X.X. t=0 0. m=audio 18120 RTP/AVP 0 110 101. a=rtpmap:0 PCMU/8000. a=rtpmap:110 speex/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=sendrecv.
# U 178.79.x.x:5060 -> 94.94.X.X:1025 SIP/2.0 407 Proxy Authentication Required. Via: SIP/2.0/UDP 94.94.X.X:5060;branch=z9hG4bK69275d13;rport=1025. From: "asterisk" sip:512@test.domain;tag=as4531a4ee. To: sip:528@test.domain:5060;tag=333a0370df4a40d5d5a0c21bb156e2a6.038f. Call-ID: 2ea7e0f3173949d40b69385b62b79d6a@test.domain. CSeq: 102 INVITE. Proxy-Authenticate: Digest realm="test.domain", nonce="Ugo+XVIKPTFBd9tXC0QemfR9bHiZXO6x". Server: kamailio (4.0.2 (x86_64/linux)). Content-Length: 0. .
# U 94.94.X.X:1025 -> 178.79.x.x:5060 INVITE sip:528@test.domain:5060 SIP/2.0. Via: SIP/2.0/UDP 94.94.X.X:5060;branch=z9hG4bK69275d13;rport. Max-Forwards: 70. From: "asterisk" sip:512@test.domain;tag=as4531a4ee. To: sip:528@test.domain:5060. Contact: sip:512@94.94.X.X:5060. Call-ID: 2ea7e0f3173949d40b69385b62b79d6a@test.domain. CSeq: 102 INVITE. User-Agent: DICE 1.8.20.1. Date: Tue, 13 Aug 2013 14:04:11 GMT. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH. Supported: replaces, timer. Content-Type: application/sdp. Content-Length: 262. . v=0. o=root 883039875 883039875 IN IP4 94.94.X.X. s=Asterisk PBX 11.3.0. c=IN IP4 94.94.X.X. t=0 0. m=audio 18120 RTP/AVP 0 110 101. a=rtpmap:0 PCMU/8000. a=rtpmap:110 speex/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=sendrecv.
# U 178.79.x.x:5060 -> 94.94.X.X:1025 SIP/2.0 407 Proxy Authentication Required. Via: SIP/2.0/UDP 94.94.X.X:5060;branch=z9hG4bK69275d13;rport=1025. From: "asterisk" sip:512@test.domain;tag=as4531a4ee. To: sip:528@test.domain:5060;tag=333a0370df4a40d5d5a0c21bb156e2a6.038f. Call-ID: 2ea7e0f3173949d40b69385b62b79d6a@test.domain. CSeq: 102 INVITE. Proxy-Authenticate: Digest realm="test.domain", nonce="Ugo+XVIKPTFBd9tXC0QemfR9bHiZXO6x". Server: kamailio (4.0.2 (x86_64/linux)). Content-Length: 0. .
# U 94.94.X.X:1025 -> 178.79.x.x:5060 ACK sip:528@test.domain:5060 SIP/2.0. Via: SIP/2.0/UDP 94.94.X.X:5060;branch=z9hG4bK69275d13;rport. Max-Forwards: 70. From: "asterisk" sip:512@test.domain;tag=as4531a4ee. To: sip:528@test.domain:5060;tag=333a0370df4a40d5d5a0c21bb156e2a6.038f. Contact: sip:512@94.94.X.X:5060. Call-ID: 2ea7e0f3173949d40b69385b62b79d6a@test.domain. CSeq: 102 ACK. User-Agent: DICE 1.8.20.1. Content-Length: 0. .
# U 94.94.X.X:1025 -> 178.79.x.x:5060 ACK sip:528@test.domain:5060 SIP/2.0. Via: SIP/2.0/UDP 94.94.X.X:5060;branch=z9hG4bK69275d13;rport. Max-Forwards: 70. From: "asterisk" sip:512@test.domain;tag=as4531a4ee. To: sip:528@test.domain:5060. Contact: sip:512@94.94.X.X:5060. Call-ID: 2ea7e0f3173949d40b69385b62b79d6a@test.domain. CSeq: 102 ACK. User-Agent: DICE 1.8.20.1. Content-Length: 0. .
------------------ This is the client behind NAT ---------------------------
REGISTER sip:test.domain;transport=tcp;hide SIP/2.0 Via: SIP/2.0/TCP 192.168.2.90:49519;rport;branch=z9hG4bKPj92d04d45-bd84-45b3-9439-563ebfaebf00;alias Max-Forwards: 70 From: sip:528@test.domain;hide;tag=ab763476-0fdb-4d85-9064-1deabcaa23e4 To: sip:528@test.domain;hide Call-ID: 3ebadaeb-e6d6-42e3-9362-5ca4b5ddd8e1 CSeq: 30028 REGISTER User-Agent: PJSUA v2.1-svn Linux-3.10.5.201/x86_64/glibc-2.17 Contact: sip:528@192.168.2.90:5060;transport=TCP;ob Expires: 300 Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Content-Length: 0
--end msg-- 15:54:02.580 pjsua_acc.c .Acc 0: Registration sent
15:54:02.681 tcpc0x1515a08 !TCP transport 192.168.2.90:49519 is connected to 178.79.x.x:5060
15:54:02.681 pjsua_app.c SIP TCP transport is connected to [178.79.x.x:5060] 15:54:02.785 pjsua_core.c .RX 538 bytes Response msg 401/REGISTER/cseq=30028 (rdata0x1515cf8) from TCP 178.79.x.x:5060: SIP/2.0 401 Unauthorized Via: SIP/2.0/TCP 192.168.2.90:49519;rport=49519;branch=z9hG4bKPj92d04d45-bd84-45b3-9439-563ebfaebf00;alias From: sip:528@test.domain;hide;tag=ab763476-0fdb-4d85-9064-1deabcaa23e4 To: sip:528@test.domain;hide;tag=333a0370df4a40d5d5a0c21bb156e2a6.9663 Call-ID: 3ebadaeb-e6d6-42e3-9362-5ca4b5ddd8e1 CSeq: 30028 REGISTER WWW-Authenticate: Digest realm="test.domain", nonce="Ugo7/FIKOtBneWAQzD/VoJuucFT3cNuF" Server: kamailio (4.0.2 (x86_64/linux)) Content-Length: 0
--end msg-- 15:54:02.785 pjsua_core.c ....TX 849 bytes Request msg REGISTER/cseq=30029 (tdta0x1513000) to TCP 178.79.x.x:5060: REGISTER sip:test.domain;transport=tcp;hide SIP/2.0 Via: SIP/2.0/TCP 192.168.2.90:49519;rport;branch=z9hG4bKPj03a4a7eb-d086-4add-b84c-ca5c6cf4d05d;alias Max-Forwards: 70 From: sip:528@test.domain;hide;tag=ab763476-0fdb-4d85-9064-1deabcaa23e4 To: sip:528@test.domain;hide Call-ID: 3ebadaeb-e6d6-42e3-9362-5ca4b5ddd8e1 CSeq: 30029 REGISTER User-Agent: PJSUA v2.1-svn Linux-3.10.5.201/x86_64/glibc-2.17 Contact: sip:528@192.168.2.90:5060;transport=TCP;ob Expires: 300 Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Authorization: Digest username="528", realm="test.domain", nonce="Ugo7/FIKOtBneWAQzD/VoJuucFT3cNuF", uri="sip:test.domain;transport=tcp;hide", response="c3dd687602ec35b2403b3eb142b496f5" Content-Length: 0
--end msg-- 15:54:02.890 pjsua_core.c .RX 495 bytes Response msg 200/REGISTER/cseq=30029 (rdata0x1515cf8) from TCP 178.79.x.x:5060: SIP/2.0 200 OK Via: SIP/2.0/TCP 192.168.2.90:49519;rport=49519;branch=z9hG4bKPj03a4a7eb-d086-4add-b84c-ca5c6cf4d05d;alias From: sip:528@test.domain;hide;tag=ab763476-0fdb-4d85-9064-1deabcaa23e4 To: sip:528@test.domain;hide;tag=333a0370df4a40d5d5a0c21bb156e2a6.189b Call-ID: 3ebadaeb-e6d6-42e3-9362-5ca4b5ddd8e1 CSeq: 30029 REGISTER Contact: sip:528@192.168.2.90:5060;transport=TCP;ob;expires=300 Server: kamailio (4.0.2 (x86_64/linux)) Content-Length: 0
--end msg-- 15:54:02.890 pjsua_acc.c ....SIP outbound status for acc 0 is not active 15:54:02.890 pjsua_acc.c ....sip:528@test.domain;transport=tcp;hide: registration success, status=200 (OK), will re-register in 300 seconds
---------------------------------------------------
Daniel
On 8/13/13 3:23 PM, Roberto Fichera wrote:
On 08/13/2013 03:15 PM, Roberto Fichera wrote:
On 08/13/2013 02:33 PM, Daniel-Constantin Mierla wrote:
Hello,
On 8/13/13 1:10 PM, Roberto Fichera wrote:
On 08/13/2013 12:03 PM, Daniel-Constantin Mierla wrote: > Hello, > > you should grab the ngrep for such call to understand better what happens. Also, dumping the location records > will be > useful (kamctl ul show). > > Also, be sure that tcp connection lifetime is long enough to survive re-registration. To avoid trying to open > connections behind nat, use set_forward_no_connect() for calls involving nat traversal. I'm using the default conf coming from fedora rpm. So, mainly the problem seems related to kamailio which doesn't reuse the TCP port used by NATed clients. I've also notice that the received field isn't set at all, so this means that the contact will not get aliased at all.
I would really like to have a look to a working cfg file for TCP NATed clients that reuse the TCP port. Even better if the configuration is based on the fedora default rpm.
if received is not set, then means the register was not detected as coming from behind nat. Is the phone using stun?
I'm testing with a normal rtpproxy configuration. BTW udp -> udp work perfectly.
Again, put here the ngrep for registration and a call to see if something is wrong with signaling. There is no help that we can provide otherwise. The default config works fine for tcp and natted clients, I use it everywhere for this case without issues.
I tried the default cfg enabling both NAT and RTPproxy, but seems that kamailio doesn't reuse TCP ports. Anyway, this is a call from UDP (512) -> TCP (526) both behind the same NAT, from kamailio point of view
I forgot to say that the received field is now present because I've changed the route[NATDETECT] in the default configuration as
route[NATDETECT] { #!ifdef WITH_NAT force_rport();
-->>> if (nat_uac_test("19") || proto != UDP) { if (is_method("REGISTER")) { fix_nated_register(); } else { fix_nated_contact(); } setflag(FLT_NATS); } #!endif return; }
[root@proxy ~]# kamctl ul show 526 Contact:: sip:526@94.94.X.X:1238;transport=TCP;ob;q=;expires=537;flags=0x0;cflags=0x40;state=0;socket=tcp:178.79.x.x:5060;methods=0x1FDF;received=sip:94.94.X.X:61922;transport=TCP;user_agent=<DICE
Smartphone 1.0/iPhone>;reg-id=0 [root@proxy ~]# kamctl ul show 512 Contact:: sip:512@94.94.X.X:5060;q=;expires=32;flags=0x0;cflags=0x40;state=0;socket=udp:178.79.x.x:5060;methods=0xFFFFFFFF;received=sip:94.94.X.X:1025;user_agent=<DICE
1.8.20.1>;reg-id=0 [root@proxy ~]#
U 94.94.X.X:1025 -> 178.79.x.x:5060 INVITE sip:526@test.domain:5060 SIP/2.0. Via: SIP/2.0/UDP 94.94.X.X:5060;branch=z9hG4bK4a420cee;rport. Max-Forwards: 70. From: "asterisk" sip:512@test.domain;tag=as76007db0. To: sip:526@test.domain:5060. Contact: sip:512@94.94.X.X:5060. Call-ID: 068a5a23639785a7583d952d6f9bca84@test.domain. CSeq: 102 INVITE. User-Agent: DICE 1.8.20.1. Date: Tue, 13 Aug 2013 13:04:30 GMT. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH. Supported: replaces, timer. Content-Type: application/sdp. Content-Length: 264. . v=0. o=root 1263161426 1263161426 IN IP4 94.94.X.X. s=Asterisk PBX 11.3.0. c=IN IP4 94.94.X.X. t=0 0. m=audio 10782 RTP/AVP 0 110 101. a=rtpmap:0 PCMU/8000. a=rtpmap:110 speex/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=sendrecv.
# U 178.79.x.x:5060 -> 94.94.X.X:1025 SIP/2.0 407 Proxy Authentication Required. Via: SIP/2.0/UDP 94.94.X.X:5060;branch=z9hG4bK4a420cee;rport=1025. From: "asterisk" sip:512@test.domain;tag=as76007db0. To: sip:526@test.domain:5060;tag=333a0370df4a40d5d5a0c21bb156e2a6.4a00. Call-ID: 068a5a23639785a7583d952d6f9bca84@test.domain. CSeq: 102 INVITE. Proxy-Authenticate: Digest realm="test.domain", nonce="UgowYFIKLzQXDSUWUpy4xlhdXZ7WNjPt". Server: kamailio (4.0.2 (x86_64/linux)). Content-Length: 0. .
# U 94.94.X.X:1025 -> 178.79.x.x:5060 INVITE sip:526@test.domain:5060 SIP/2.0. Via: SIP/2.0/UDP 94.94.X.X:5060;branch=z9hG4bK4a420cee;rport. Max-Forwards: 70. From: "asterisk" sip:512@test.domain;tag=as76007db0. To: sip:526@test.domain:5060. Contact: sip:512@94.94.X.X:5060. Call-ID: 068a5a23639785a7583d952d6f9bca84@test.domain. CSeq: 102 INVITE. User-Agent: DICE 1.8.20.1. Date: Tue, 13 Aug 2013 13:04:30 GMT. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH. Supported: replaces, timer. Content-Type: application/sdp. Content-Length: 264. . v=0. o=root 1263161426 1263161426 IN IP4 94.94.X.X. s=Asterisk PBX 11.3.0. c=IN IP4 94.94.X.X. t=0 0. m=audio 10782 RTP/AVP 0 110 101. a=rtpmap:0 PCMU/8000. a=rtpmap:110 speex/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=sendrecv.
# U 178.79.x.x:5060 -> 94.94.X.X:1025 SIP/2.0 407 Proxy Authentication Required. Via: SIP/2.0/UDP 94.94.X.X:5060;branch=z9hG4bK4a420cee;rport=1025. From: "asterisk" sip:512@test.domain;tag=as76007db0. To: sip:526@test.domain:5060;tag=333a0370df4a40d5d5a0c21bb156e2a6.4a00. Call-ID: 068a5a23639785a7583d952d6f9bca84@test.domain. CSeq: 102 INVITE. Proxy-Authenticate: Digest realm="test.domain", nonce="UgowYFIKLzQXDSUWUpy4xlhdXZ7WNjPt". Server: kamailio (4.0.2 (x86_64/linux)). Content-Length: 0. .
# U 94.94.X.X:1025 -> 178.79.x.x:5060 ACK sip:526@test.domain:5060 SIP/2.0. Via: SIP/2.0/UDP 94.94.X.X:5060;branch=z9hG4bK4a420cee;rport. Max-Forwards: 70. From: "asterisk" sip:512@test.domain;tag=as76007db0. To: sip:526@test.domain:5060;tag=333a0370df4a40d5d5a0c21bb156e2a6.4a00. Contact: sip:512@94.94.X.X:5060. Call-ID: 068a5a23639785a7583d952d6f9bca84@test.domain. CSeq: 102 ACK. User-Agent: DICE 1.8.20.1. Content-Length: 0. .
# U 94.94.X.X:1025 -> 178.79.x.x:5060 ACK sip:526@test.domain:5060 SIP/2.0. Via: SIP/2.0/UDP 94.94.X.X:5060;branch=z9hG4bK4a420cee;rport. Max-Forwards: 70. From: "asterisk" sip:512@test.domain;tag=as76007db0. To: sip:526@test.domain:5060. Contact: sip:512@94.94.X.X:5060. Call-ID: 068a5a23639785a7583d952d6f9bca84@test.domain. CSeq: 102 ACK. User-Agent: DICE 1.8.20.1. Content-Length: 0. .
# T 178.79.x.x:5060 -> 94.94.X.X:61922 [AP] ACK sip:526@94.94.X.X:1238;transport=TCP;ob SIP/2.0. Via: SIP/2.0/TCP 178.79.x.x;branch=z9hG4bKcydzigwkX. Via: SIP/2.0/UDP 94.94.X.X:5060;branch=z9hG4bK4a420cee;rport=1025. Max-Forwards: 16. From: "asterisk" sip:512@test.domain;tag=as76007db0. To: sip:526@test.domain:5060. Contact: sip:512@94.94.X.X:1025. Call-ID: 068a5a23639785a7583d952d6f9bca84@test.domain. CSeq: 102 ACK. User-Agent: DICE 1.8.20.1. Content-Length: 0. .
# T 178.79.x.x:5060 -> 94.94.X.X:61922 [AP] ACK sip:526@94.94.X.X:1238;transport=TCP;ob SIP/2.0. Via: SIP/2.0/TCP 178.79.x.x;branch=z9hG4bKcydzigwkX. Via: SIP/2.0/UDP 94.94.X.X:5060;branch=z9hG4bK4a420cee;rport=1025. Max-Forwards: 16. From: "asterisk" sip:512@test.domain;tag=as76007db0. To: sip:526@test.domain:5060. Contact: sip:512@94.94.X.X:1025. Call-ID: 068a5a23639785a7583d952d6f9bca84@test.domain. CSeq: 102 ACK. User-Agent: DICE 1.8.20.1. Content-Length: 0. .
Cheers, Daniel
Cheers, Roberto Fichera.
> Cheers, > Daniel > > On 7/30/13 6:44 PM, Roberto Fichera wrote: >> Hi All, >> >> Sorry for cross-posting this email to PJLIB, but maybe there are some things related. >> Anyhow! I'm having problems on kamailio v4.0.2 under Fedora 18 64bit and TCP client like iPhone using PJSIP as SIP >> library. >> Basically once the iPhone side in close the call (TCP->UDP) I'm getting the error below. Kamailio is running under >> a VPS >> without >> NATed network so it uses a real public address. Furthermore, note that tcp_main is answering to a 192.168.2.98 ip >> address >> which is the iPhone client. This looks really strange to me since it should answer directly to the public/port used >> for >> the registration >> and not to a such kind of reserved address. The kamilio configuration is basically the default with a very few >> changes >> like NAT, rtpproxy and postgresql backend. >> >> This problems doesn't happen at all when using UDP->UDP calls. But I cannot use it because as you certain know UDP >> connection under iPhone will not work when the application run in background mode. >> >> Can someone suggest how to solve this issue or maybe suggest a TCP working solution for iPhone? >> >> Thanks in advance. >> Roberto Fichera. >> >> Jul 30 16:21:53 proxy /usr/sbin/kamailio[9502]: ERROR: <core> [tcp_main.c:4432]: tcpconn_main_timeout(): connect >> 192.168.2.98:5060 failed (timeout) >> Jul 30 16:21:55 proxy rtpproxy[2262]: DBUG:get_command: received command "9483_9 D >> 12d1d19926c4ff742a52f0c855b1bb83@94.94.x.x:5060 as74e0c388 GROahimCK6KTrl5CkYEg7nuoPIIXZ8cj" >> Jul 30 16:21:55 proxy rtpproxy[2262]: INFO:handle_delete: forcefully deleting session 1 on ports 15604/17354 >> Jul 30 16:21:55 proxy rtpproxy[2262]: INFO:remove_session: RTP stats: 354 in from callee, 603 in from caller, 957 >> relayed, 0 dropped >> Jul 30 16:21:55 proxy rtpproxy[2262]: INFO:remove_session: RTCP stats: 5 in from callee, 2 in from caller, 7 >> relayed, 0 >> dropped >> Jul 30 16:21:55 proxy rtpproxy[2262]: INFO:remove_session: session on ports 15604/17354 is cleaned up >> Jul 30 16:21:55 proxy rtpproxy[2262]: DBUG:doreply: sending reply "9483_9 0 >> Jul 30 16:21:55 proxy rtpproxy[2262]: " >> Jul 30 16:22:04 proxy /usr/sbin/kamailio[9502]: ERROR: <core> [tcp_main.c:4432]: tcpconn_main_timeout(): connect >> 192.168.2.98:5060 failed (timeout) >> Jul 30 16:22:14 proxy rtpproxy[2262]: DBUG:get_command: received command "9496_16 D >> 12d1d19926c4ff742a52f0c855b1bb83@94.94.x.x:5060 GROahimCK6KTrl5CkYEg7nuoPIIXZ8cj as74e0c388" >> Jul 30 16:22:14 proxy rtpproxy[2262]: INFO:handle_command: delete request failed: session >> 12d1d19926c4ff742a52f0c855b1bb83@94.94.x.x:5060, tags GROahimCK6KTrl5CkYEg7nuoPIIXZ8cj/as74e0c388 not found >> Jul 30 16:22:14 proxy rtpproxy[2262]: DBUG:doreply: sending reply "9496_16 E8 >> Jul 30 16:22:14 proxy rtpproxy[2262]: " >> >> >> _______________________________________________ >> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >> sr-users@lists.sip-router.org >> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Hello,
as you can see in the REGISTER, the phone give a public IP where it can be contacted and kamailio tries to deliver to that address sip:528@94.94.X.X:1274;transport=TCP
REGISTER comes from another port, but that is allowed in SIP.
You should disable stun in the client and let the server care of nat traversal.
Or you can try to add flag 128 to nat test: - http://kamailio.org/docs/modules/stable/modules/nathelper.html#idp15376536
Also, you can try to add an alias for contact port: - http://www.kamailio.org/wiki/cookbooks/4.0.x/core#force_tcp_alias
Cheers, Daniel
On 8/13/13 4:17 PM, Roberto Fichera wrote:
On 08/13/2013 03:32 PM, Roberto Fichera wrote:
On 08/13/2013 03:25 PM, Daniel-Constantin Mierla wrote:
Can you get a ngrep trace for a registration as well (for the phone using tcp)?
Ok! I'll use pjsua from my local machine connecting in the same way as the TCP client was doing. The TCP client it's an iPhone using the same pjlib library.
I can confirm that the default cfg for TCP client doesn't work for me. My cfg file is attached. The TCP client doesn't receive any package at INVITE. Finally in /var/log/message I get this log below:
Aug 13 14:05:37 proxy /usr/sbin/kamailio[8401]: ERROR: <core> [tcp_main.c:4247]: handle_tcpconn_ev(): connect 94.94.X.X:1274 failed
Contact:: sip:528@94.94.X.X:1274;transport=TCP;ob;q=;expires=38;flags=0x0;cflags=0x0;state=0;socket=tcp:178.79.X.X:5060;methods=0x1FDF;user_agent=<PJSUA v2.1-svn Linux-3.10.5.201/x86_64/glibc-2.17>;reg-id=0 [root@proxy ~]# kamctl ul show 512 Contact:: sip:512@94.94.X.X:5060;q=;expires=61;flags=0x0;cflags=0x40;state=0;socket=udp:178.79.X.X:5060;methods=0xFFFFFFFF;received=sip:94.94.X.X:1025 user_agent=<DICE 1.8.20.1>;reg-id=0
[root@proxy ~]# ngrep -W byline -d eth0 port 5060 interface: eth0 (178.79.X.X/255.255.255.0) filter: (ip or ip6) and ( port 5060 ) # T 94.94.X.X:49519 -> 178.79.x.x:5060 [AP] REGISTER sip:test.domain;transport=tcp;hide SIP/2.0. Via: SIP/2.0/TCP 94.94.X.X:49519;rport;branch=z9hG4bKPje03c11fd-d742-4502-98d8-69ca456ddd56;alias. Max-Forwards: 70. From: sip:528@test.domain;hide;tag=64c9e9b9-cb23-4a79-bb0d-2497135e449b. To: sip:528@test.domain;hide. Call-ID: 3ebadaeb-e6d6-42e3-9362-5ca4b5ddd8e1. CSeq: 30034 REGISTER. User-Agent: PJSUA v2.1-svn Linux-3.10.5.201/x86_64/glibc-2.17. Contact: sip:528@94.94.X.X:1274;transport=TCP;ob. Expires: 300. Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS. Content-Length: 0. .
# T 178.79.x.x:5060 -> 94.94.X.X:49519 [AP] SIP/2.0 401 Unauthorized. Via: SIP/2.0/TCP 94.94.X.X:49519;rport=49519;branch=z9hG4bKPje03c11fd-d742-4502-98d8-69ca456ddd56;alias. From: sip:528@test.domain;hide;tag=64c9e9b9-cb23-4a79-bb0d-2497135e449b. To: sip:528@test.domain;hide;tag=333a0370df4a40d5d5a0c21bb156e2a6.9167. Call-ID: 3ebadaeb-e6d6-42e3-9362-5ca4b5ddd8e1. CSeq: 30034 REGISTER. WWW-Authenticate: Digest realm="test.domain", nonce="Ugo+VFIKPSgVy8fh9fSy9SDLvT0wO4QV". Server: kamailio (4.0.2 (x86_64/linux)). Content-Length: 0. .
# U 178.79.x.x:5060 -> 94.94.X.X:1025 .... ## T 94.94.X.X:49519 -> 178.79.x.x:5060 [AP] REGISTER sip:test.domain;transport=tcp;hide SIP/2.0. Via: SIP/2.0/TCP 94.94.X.X:49519;rport;branch=z9hG4bKPj68a9c487-ae8c-428a-8c50-dd195a46a6bf;alias. Max-Forwards: 70. From: sip:528@test.domain;hide;tag=64c9e9b9-cb23-4a79-bb0d-2497135e449b. To: sip:528@test.domain;hide. Call-ID: 3ebadaeb-e6d6-42e3-9362-5ca4b5ddd8e1. CSeq: 30035 REGISTER. User-Agent: PJSUA v2.1-svn Linux-3.10.5.201/x86_64/glibc-2.17. Contact: sip:528@94.94.X.X:1274;transport=TCP;ob. Expires: 300. Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS. Authorization: Digest username="528", realm="test.domain", nonce="Ugo+VFIKPSgVy8fh9fSy9SDLvT0wO4QV", uri="sip:test.domain;transport=tcp;hide", response="ac1c7311ccb887fc8fb494d8ebf1bd36". Content-Length: 0. .
# T 178.79.x.x:5060 -> 94.94.X.X:49519 [AP] SIP/2.0 200 OK. Via: SIP/2.0/TCP 94.94.X.X:49519;rport=49519;branch=z9hG4bKPj68a9c487-ae8c-428a-8c50-dd195a46a6bf;alias. From: sip:528@test.domain;hide;tag=64c9e9b9-cb23-4a79-bb0d-2497135e449b. To: sip:528@test.domain;hide;tag=333a0370df4a40d5d5a0c21bb156e2a6.5440. Call-ID: 3ebadaeb-e6d6-42e3-9362-5ca4b5ddd8e1. CSeq: 30035 REGISTER. Contact: sip:528@94.94.X.X:1274;transport=TCP;ob;expires=300. Server: kamailio (4.0.2 (x86_64/linux)). Content-Length: 0. .
## U 94.94.X.X:1025 -> 178.79.x.x:5060 INVITE sip:528@test.domain:5060 SIP/2.0. Via: SIP/2.0/UDP 94.94.X.X:5060;branch=z9hG4bK69275d13;rport. Max-Forwards: 70. From: "asterisk" sip:512@test.domain;tag=as4531a4ee. To: sip:528@test.domain:5060. Contact: sip:512@94.94.X.X:5060. Call-ID: 2ea7e0f3173949d40b69385b62b79d6a@test.domain. CSeq: 102 INVITE. User-Agent: DICE 1.8.20.1. Date: Tue, 13 Aug 2013 14:04:11 GMT. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH. Supported: replaces, timer. Content-Type: application/sdp. Content-Length: 262. . v=0. o=root 883039875 883039875 IN IP4 94.94.X.X. s=Asterisk PBX 11.3.0. c=IN IP4 94.94.X.X. t=0 0. m=audio 18120 RTP/AVP 0 110 101. a=rtpmap:0 PCMU/8000. a=rtpmap:110 speex/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=sendrecv.
# U 178.79.x.x:5060 -> 94.94.X.X:1025 SIP/2.0 407 Proxy Authentication Required. Via: SIP/2.0/UDP 94.94.X.X:5060;branch=z9hG4bK69275d13;rport=1025. From: "asterisk" sip:512@test.domain;tag=as4531a4ee. To: sip:528@test.domain:5060;tag=333a0370df4a40d5d5a0c21bb156e2a6.038f. Call-ID: 2ea7e0f3173949d40b69385b62b79d6a@test.domain. CSeq: 102 INVITE. Proxy-Authenticate: Digest realm="test.domain", nonce="Ugo+XVIKPTFBd9tXC0QemfR9bHiZXO6x". Server: kamailio (4.0.2 (x86_64/linux)). Content-Length: 0. .
# U 94.94.X.X:1025 -> 178.79.x.x:5060 INVITE sip:528@test.domain:5060 SIP/2.0. Via: SIP/2.0/UDP 94.94.X.X:5060;branch=z9hG4bK69275d13;rport. Max-Forwards: 70. From: "asterisk" sip:512@test.domain;tag=as4531a4ee. To: sip:528@test.domain:5060. Contact: sip:512@94.94.X.X:5060. Call-ID: 2ea7e0f3173949d40b69385b62b79d6a@test.domain. CSeq: 102 INVITE. User-Agent: DICE 1.8.20.1. Date: Tue, 13 Aug 2013 14:04:11 GMT. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH. Supported: replaces, timer. Content-Type: application/sdp. Content-Length: 262. . v=0. o=root 883039875 883039875 IN IP4 94.94.X.X. s=Asterisk PBX 11.3.0. c=IN IP4 94.94.X.X. t=0 0. m=audio 18120 RTP/AVP 0 110 101. a=rtpmap:0 PCMU/8000. a=rtpmap:110 speex/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=sendrecv.
# U 178.79.x.x:5060 -> 94.94.X.X:1025 SIP/2.0 407 Proxy Authentication Required. Via: SIP/2.0/UDP 94.94.X.X:5060;branch=z9hG4bK69275d13;rport=1025. From: "asterisk" sip:512@test.domain;tag=as4531a4ee. To: sip:528@test.domain:5060;tag=333a0370df4a40d5d5a0c21bb156e2a6.038f. Call-ID: 2ea7e0f3173949d40b69385b62b79d6a@test.domain. CSeq: 102 INVITE. Proxy-Authenticate: Digest realm="test.domain", nonce="Ugo+XVIKPTFBd9tXC0QemfR9bHiZXO6x". Server: kamailio (4.0.2 (x86_64/linux)). Content-Length: 0. .
# U 94.94.X.X:1025 -> 178.79.x.x:5060 ACK sip:528@test.domain:5060 SIP/2.0. Via: SIP/2.0/UDP 94.94.X.X:5060;branch=z9hG4bK69275d13;rport. Max-Forwards: 70. From: "asterisk" sip:512@test.domain;tag=as4531a4ee. To: sip:528@test.domain:5060;tag=333a0370df4a40d5d5a0c21bb156e2a6.038f. Contact: sip:512@94.94.X.X:5060. Call-ID: 2ea7e0f3173949d40b69385b62b79d6a@test.domain. CSeq: 102 ACK. User-Agent: DICE 1.8.20.1. Content-Length: 0. .
# U 94.94.X.X:1025 -> 178.79.x.x:5060 ACK sip:528@test.domain:5060 SIP/2.0. Via: SIP/2.0/UDP 94.94.X.X:5060;branch=z9hG4bK69275d13;rport. Max-Forwards: 70. From: "asterisk" sip:512@test.domain;tag=as4531a4ee. To: sip:528@test.domain:5060. Contact: sip:512@94.94.X.X:5060. Call-ID: 2ea7e0f3173949d40b69385b62b79d6a@test.domain. CSeq: 102 ACK. User-Agent: DICE 1.8.20.1. Content-Length: 0. .
------------------ This is the client behind NAT ---------------------------
REGISTER sip:test.domain;transport=tcp;hide SIP/2.0 Via: SIP/2.0/TCP 192.168.2.90:49519;rport;branch=z9hG4bKPj92d04d45-bd84-45b3-9439-563ebfaebf00;alias Max-Forwards: 70 From: sip:528@test.domain;hide;tag=ab763476-0fdb-4d85-9064-1deabcaa23e4 To: sip:528@test.domain;hide Call-ID: 3ebadaeb-e6d6-42e3-9362-5ca4b5ddd8e1 CSeq: 30028 REGISTER User-Agent: PJSUA v2.1-svn Linux-3.10.5.201/x86_64/glibc-2.17 Contact: sip:528@192.168.2.90:5060;transport=TCP;ob Expires: 300 Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Content-Length: 0
--end msg-- 15:54:02.580 pjsua_acc.c .Acc 0: Registration sent
15:54:02.681 tcpc0x1515a08 !TCP transport 192.168.2.90:49519 is connected to 178.79.x.x:5060
15:54:02.681 pjsua_app.c SIP TCP transport is connected to [178.79.x.x:5060] 15:54:02.785 pjsua_core.c .RX 538 bytes Response msg 401/REGISTER/cseq=30028 (rdata0x1515cf8) from TCP 178.79.x.x:5060: SIP/2.0 401 Unauthorized Via: SIP/2.0/TCP 192.168.2.90:49519;rport=49519;branch=z9hG4bKPj92d04d45-bd84-45b3-9439-563ebfaebf00;alias From: sip:528@test.domain;hide;tag=ab763476-0fdb-4d85-9064-1deabcaa23e4 To: sip:528@test.domain;hide;tag=333a0370df4a40d5d5a0c21bb156e2a6.9663 Call-ID: 3ebadaeb-e6d6-42e3-9362-5ca4b5ddd8e1 CSeq: 30028 REGISTER WWW-Authenticate: Digest realm="test.domain", nonce="Ugo7/FIKOtBneWAQzD/VoJuucFT3cNuF" Server: kamailio (4.0.2 (x86_64/linux)) Content-Length: 0
--end msg-- 15:54:02.785 pjsua_core.c ....TX 849 bytes Request msg REGISTER/cseq=30029 (tdta0x1513000) to TCP 178.79.x.x:5060: REGISTER sip:test.domain;transport=tcp;hide SIP/2.0 Via: SIP/2.0/TCP 192.168.2.90:49519;rport;branch=z9hG4bKPj03a4a7eb-d086-4add-b84c-ca5c6cf4d05d;alias Max-Forwards: 70 From: sip:528@test.domain;hide;tag=ab763476-0fdb-4d85-9064-1deabcaa23e4 To: sip:528@test.domain;hide Call-ID: 3ebadaeb-e6d6-42e3-9362-5ca4b5ddd8e1 CSeq: 30029 REGISTER User-Agent: PJSUA v2.1-svn Linux-3.10.5.201/x86_64/glibc-2.17 Contact: sip:528@192.168.2.90:5060;transport=TCP;ob Expires: 300 Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Authorization: Digest username="528", realm="test.domain", nonce="Ugo7/FIKOtBneWAQzD/VoJuucFT3cNuF", uri="sip:test.domain;transport=tcp;hide", response="c3dd687602ec35b2403b3eb142b496f5" Content-Length: 0
--end msg-- 15:54:02.890 pjsua_core.c .RX 495 bytes Response msg 200/REGISTER/cseq=30029 (rdata0x1515cf8) from TCP 178.79.x.x:5060: SIP/2.0 200 OK Via: SIP/2.0/TCP 192.168.2.90:49519;rport=49519;branch=z9hG4bKPj03a4a7eb-d086-4add-b84c-ca5c6cf4d05d;alias From: sip:528@test.domain;hide;tag=ab763476-0fdb-4d85-9064-1deabcaa23e4 To: sip:528@test.domain;hide;tag=333a0370df4a40d5d5a0c21bb156e2a6.189b Call-ID: 3ebadaeb-e6d6-42e3-9362-5ca4b5ddd8e1 CSeq: 30029 REGISTER Contact: sip:528@192.168.2.90:5060;transport=TCP;ob;expires=300 Server: kamailio (4.0.2 (x86_64/linux)) Content-Length: 0
--end msg-- 15:54:02.890 pjsua_acc.c ....SIP outbound status for acc 0 is not active 15:54:02.890 pjsua_acc.c ....sip:528@test.domain;transport=tcp;hide: registration success, status=200 (OK), will re-register in 300 seconds
Daniel
On 8/13/13 3:23 PM, Roberto Fichera wrote:
On 08/13/2013 03:15 PM, Roberto Fichera wrote:
On 08/13/2013 02:33 PM, Daniel-Constantin Mierla wrote:
Hello,
On 8/13/13 1:10 PM, Roberto Fichera wrote: > On 08/13/2013 12:03 PM, Daniel-Constantin Mierla wrote: >> Hello, >> >> you should grab the ngrep for such call to understand better what happens. Also, dumping the location records >> will be >> useful (kamctl ul show). >> >> Also, be sure that tcp connection lifetime is long enough to survive re-registration. To avoid trying to open >> connections behind nat, use set_forward_no_connect() for calls involving nat traversal. > I'm using the default conf coming from fedora rpm. So, mainly the problem seems related to kamailio > which doesn't reuse the TCP port used by NATed clients. I've also notice that the received > field isn't set at all, so this means that the contact will not get aliased at all. > > I would really like to have a look to a working cfg file for TCP NATed clients that reuse the TCP port. > Even better if the configuration is based on the fedora default rpm. if received is not set, then means the register was not detected as coming from behind nat. Is the phone using stun?
I'm testing with a normal rtpproxy configuration. BTW udp -> udp work perfectly.
Again, put here the ngrep for registration and a call to see if something is wrong with signaling. There is no help that we can provide otherwise. The default config works fine for tcp and natted clients, I use it everywhere for this case without issues.
I tried the default cfg enabling both NAT and RTPproxy, but seems that kamailio doesn't reuse TCP ports. Anyway, this is a call from UDP (512) -> TCP (526) both behind the same NAT, from kamailio point of view
I forgot to say that the received field is now present because I've changed the route[NATDETECT] in the default configuration as
route[NATDETECT] { #!ifdef WITH_NAT force_rport();
-->>> if (nat_uac_test("19") || proto != UDP) { if (is_method("REGISTER")) { fix_nated_register(); } else { fix_nated_contact(); } setflag(FLT_NATS); } #!endif return; }
[root@proxy ~]# kamctl ul show 526 Contact:: sip:526@94.94.X.X:1238;transport=TCP;ob;q=;expires=537;flags=0x0;cflags=0x40;state=0;socket=tcp:178.79.x.x:5060;methods=0x1FDF;received=sip:94.94.X.X:61922;transport=TCP;user_agent=<DICE
Smartphone 1.0/iPhone>;reg-id=0 [root@proxy ~]# kamctl ul show 512 Contact:: sip:512@94.94.X.X:5060;q=;expires=32;flags=0x0;cflags=0x40;state=0;socket=udp:178.79.x.x:5060;methods=0xFFFFFFFF;received=sip:94.94.X.X:1025;user_agent=<DICE
1.8.20.1>;reg-id=0 [root@proxy ~]#
U 94.94.X.X:1025 -> 178.79.x.x:5060 INVITE sip:526@test.domain:5060 SIP/2.0. Via: SIP/2.0/UDP 94.94.X.X:5060;branch=z9hG4bK4a420cee;rport. Max-Forwards: 70. From: "asterisk" sip:512@test.domain;tag=as76007db0. To: sip:526@test.domain:5060. Contact: sip:512@94.94.X.X:5060. Call-ID: 068a5a23639785a7583d952d6f9bca84@test.domain. CSeq: 102 INVITE. User-Agent: DICE 1.8.20.1. Date: Tue, 13 Aug 2013 13:04:30 GMT. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH. Supported: replaces, timer. Content-Type: application/sdp. Content-Length: 264. . v=0. o=root 1263161426 1263161426 IN IP4 94.94.X.X. s=Asterisk PBX 11.3.0. c=IN IP4 94.94.X.X. t=0 0. m=audio 10782 RTP/AVP 0 110 101. a=rtpmap:0 PCMU/8000. a=rtpmap:110 speex/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=sendrecv.
# U 178.79.x.x:5060 -> 94.94.X.X:1025 SIP/2.0 407 Proxy Authentication Required. Via: SIP/2.0/UDP 94.94.X.X:5060;branch=z9hG4bK4a420cee;rport=1025. From: "asterisk" sip:512@test.domain;tag=as76007db0. To: sip:526@test.domain:5060;tag=333a0370df4a40d5d5a0c21bb156e2a6.4a00. Call-ID: 068a5a23639785a7583d952d6f9bca84@test.domain. CSeq: 102 INVITE. Proxy-Authenticate: Digest realm="test.domain", nonce="UgowYFIKLzQXDSUWUpy4xlhdXZ7WNjPt". Server: kamailio (4.0.2 (x86_64/linux)). Content-Length: 0. .
# U 94.94.X.X:1025 -> 178.79.x.x:5060 INVITE sip:526@test.domain:5060 SIP/2.0. Via: SIP/2.0/UDP 94.94.X.X:5060;branch=z9hG4bK4a420cee;rport. Max-Forwards: 70. From: "asterisk" sip:512@test.domain;tag=as76007db0. To: sip:526@test.domain:5060. Contact: sip:512@94.94.X.X:5060. Call-ID: 068a5a23639785a7583d952d6f9bca84@test.domain. CSeq: 102 INVITE. User-Agent: DICE 1.8.20.1. Date: Tue, 13 Aug 2013 13:04:30 GMT. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH. Supported: replaces, timer. Content-Type: application/sdp. Content-Length: 264. . v=0. o=root 1263161426 1263161426 IN IP4 94.94.X.X. s=Asterisk PBX 11.3.0. c=IN IP4 94.94.X.X. t=0 0. m=audio 10782 RTP/AVP 0 110 101. a=rtpmap:0 PCMU/8000. a=rtpmap:110 speex/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=sendrecv.
# U 178.79.x.x:5060 -> 94.94.X.X:1025 SIP/2.0 407 Proxy Authentication Required. Via: SIP/2.0/UDP 94.94.X.X:5060;branch=z9hG4bK4a420cee;rport=1025. From: "asterisk" sip:512@test.domain;tag=as76007db0. To: sip:526@test.domain:5060;tag=333a0370df4a40d5d5a0c21bb156e2a6.4a00. Call-ID: 068a5a23639785a7583d952d6f9bca84@test.domain. CSeq: 102 INVITE. Proxy-Authenticate: Digest realm="test.domain", nonce="UgowYFIKLzQXDSUWUpy4xlhdXZ7WNjPt". Server: kamailio (4.0.2 (x86_64/linux)). Content-Length: 0. .
# U 94.94.X.X:1025 -> 178.79.x.x:5060 ACK sip:526@test.domain:5060 SIP/2.0. Via: SIP/2.0/UDP 94.94.X.X:5060;branch=z9hG4bK4a420cee;rport. Max-Forwards: 70. From: "asterisk" sip:512@test.domain;tag=as76007db0. To: sip:526@test.domain:5060;tag=333a0370df4a40d5d5a0c21bb156e2a6.4a00. Contact: sip:512@94.94.X.X:5060. Call-ID: 068a5a23639785a7583d952d6f9bca84@test.domain. CSeq: 102 ACK. User-Agent: DICE 1.8.20.1. Content-Length: 0. .
# U 94.94.X.X:1025 -> 178.79.x.x:5060 ACK sip:526@test.domain:5060 SIP/2.0. Via: SIP/2.0/UDP 94.94.X.X:5060;branch=z9hG4bK4a420cee;rport. Max-Forwards: 70. From: "asterisk" sip:512@test.domain;tag=as76007db0. To: sip:526@test.domain:5060. Contact: sip:512@94.94.X.X:5060. Call-ID: 068a5a23639785a7583d952d6f9bca84@test.domain. CSeq: 102 ACK. User-Agent: DICE 1.8.20.1. Content-Length: 0. .
# T 178.79.x.x:5060 -> 94.94.X.X:61922 [AP] ACK sip:526@94.94.X.X:1238;transport=TCP;ob SIP/2.0. Via: SIP/2.0/TCP 178.79.x.x;branch=z9hG4bKcydzigwkX. Via: SIP/2.0/UDP 94.94.X.X:5060;branch=z9hG4bK4a420cee;rport=1025. Max-Forwards: 16. From: "asterisk" sip:512@test.domain;tag=as76007db0. To: sip:526@test.domain:5060. Contact: sip:512@94.94.X.X:1025. Call-ID: 068a5a23639785a7583d952d6f9bca84@test.domain. CSeq: 102 ACK. User-Agent: DICE 1.8.20.1. Content-Length: 0. .
# T 178.79.x.x:5060 -> 94.94.X.X:61922 [AP] ACK sip:526@94.94.X.X:1238;transport=TCP;ob SIP/2.0. Via: SIP/2.0/TCP 178.79.x.x;branch=z9hG4bKcydzigwkX. Via: SIP/2.0/UDP 94.94.X.X:5060;branch=z9hG4bK4a420cee;rport=1025. Max-Forwards: 16. From: "asterisk" sip:512@test.domain;tag=as76007db0. To: sip:526@test.domain:5060. Contact: sip:512@94.94.X.X:1025. Call-ID: 068a5a23639785a7583d952d6f9bca84@test.domain. CSeq: 102 ACK. User-Agent: DICE 1.8.20.1. Content-Length: 0. .
Cheers, Daniel > Cheers, > Roberto Fichera. > >> Cheers, >> Daniel >> >> On 7/30/13 6:44 PM, Roberto Fichera wrote: >>> Hi All, >>> >>> Sorry for cross-posting this email to PJLIB, but maybe there are some things related. >>> Anyhow! I'm having problems on kamailio v4.0.2 under Fedora 18 64bit and TCP client like iPhone using PJSIP as SIP >>> library. >>> Basically once the iPhone side in close the call (TCP->UDP) I'm getting the error below. Kamailio is running under >>> a VPS >>> without >>> NATed network so it uses a real public address. Furthermore, note that tcp_main is answering to a 192.168.2.98 ip >>> address >>> which is the iPhone client. This looks really strange to me since it should answer directly to the public/port used >>> for >>> the registration >>> and not to a such kind of reserved address. The kamilio configuration is basically the default with a very few >>> changes >>> like NAT, rtpproxy and postgresql backend. >>> >>> This problems doesn't happen at all when using UDP->UDP calls. But I cannot use it because as you certain know UDP >>> connection under iPhone will not work when the application run in background mode. >>> >>> Can someone suggest how to solve this issue or maybe suggest a TCP working solution for iPhone? >>> >>> Thanks in advance. >>> Roberto Fichera. >>> >>> Jul 30 16:21:53 proxy /usr/sbin/kamailio[9502]: ERROR: <core> [tcp_main.c:4432]: tcpconn_main_timeout(): connect >>> 192.168.2.98:5060 failed (timeout) >>> Jul 30 16:21:55 proxy rtpproxy[2262]: DBUG:get_command: received command "9483_9 D >>> 12d1d19926c4ff742a52f0c855b1bb83@94.94.x.x:5060 as74e0c388 GROahimCK6KTrl5CkYEg7nuoPIIXZ8cj" >>> Jul 30 16:21:55 proxy rtpproxy[2262]: INFO:handle_delete: forcefully deleting session 1 on ports 15604/17354 >>> Jul 30 16:21:55 proxy rtpproxy[2262]: INFO:remove_session: RTP stats: 354 in from callee, 603 in from caller, 957 >>> relayed, 0 dropped >>> Jul 30 16:21:55 proxy rtpproxy[2262]: INFO:remove_session: RTCP stats: 5 in from callee, 2 in from caller, 7 >>> relayed, 0 >>> dropped >>> Jul 30 16:21:55 proxy rtpproxy[2262]: INFO:remove_session: session on ports 15604/17354 is cleaned up >>> Jul 30 16:21:55 proxy rtpproxy[2262]: DBUG:doreply: sending reply "9483_9 0 >>> Jul 30 16:21:55 proxy rtpproxy[2262]: " >>> Jul 30 16:22:04 proxy /usr/sbin/kamailio[9502]: ERROR: <core> [tcp_main.c:4432]: tcpconn_main_timeout(): connect >>> 192.168.2.98:5060 failed (timeout) >>> Jul 30 16:22:14 proxy rtpproxy[2262]: DBUG:get_command: received command "9496_16 D >>> 12d1d19926c4ff742a52f0c855b1bb83@94.94.x.x:5060 GROahimCK6KTrl5CkYEg7nuoPIIXZ8cj as74e0c388" >>> Jul 30 16:22:14 proxy rtpproxy[2262]: INFO:handle_command: delete request failed: session >>> 12d1d19926c4ff742a52f0c855b1bb83@94.94.x.x:5060, tags GROahimCK6KTrl5CkYEg7nuoPIIXZ8cj/as74e0c388 not found >>> Jul 30 16:22:14 proxy rtpproxy[2262]: DBUG:doreply: sending reply "9496_16 E8 >>> Jul 30 16:22:14 proxy rtpproxy[2262]: " >>> >>> >>> _______________________________________________ >>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >>> sr-users@lists.sip-router.org >>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
On 08/14/2013 10:19 AM, Daniel-Constantin Mierla wrote:
Hi,
Hello,
as you can see in the REGISTER, the phone give a public IP where it can be contacted and kamailio tries to deliver to that address sip:528@94.94.X.X:1274;transport=TCP
REGISTER comes from another port, but that is allowed in SIP.
Yep! That's the problem
You should disable stun in the client and let the server care of nat traversal.
As already said all client doesn't use any stun-
Or you can try to add flag 128 to nat test:
Also, you can try to add an alias for contact port:
Ok! I'll try both!
Cheers, Roberto Fichera.
Cheers, Daniel
On 8/13/13 4:17 PM, Roberto Fichera wrote:
On 08/13/2013 03:32 PM, Roberto Fichera wrote:
On 08/13/2013 03:25 PM, Daniel-Constantin Mierla wrote:
Can you get a ngrep trace for a registration as well (for the phone using tcp)?
Ok! I'll use pjsua from my local machine connecting in the same way as the TCP client was doing. The TCP client it's an iPhone using the same pjlib library.
I can confirm that the default cfg for TCP client doesn't work for me. My cfg file is attached. The TCP client doesn't receive any package at INVITE. Finally in /var/log/message I get this log below:
Aug 13 14:05:37 proxy /usr/sbin/kamailio[8401]: ERROR: <core> [tcp_main.c:4247]: handle_tcpconn_ev(): connect 94.94.X.X:1274 failed
Contact:: sip:528@94.94.X.X:1274;transport=TCP;ob;q=;expires=38;flags=0x0;cflags=0x0;state=0;socket=tcp:178.79.X.X:5060;methods=0x1FDF;user_agent=<PJSUA
v2.1-svn Linux-3.10.5.201/x86_64/glibc-2.17>;reg-id=0 [root@proxy ~]# kamctl ul show 512 Contact:: sip:512@94.94.X.X:5060;q=;expires=61;flags=0x0;cflags=0x40;state=0;socket=udp:178.79.X.X:5060;methods=0xFFFFFFFF;received=sip:94.94.X.X:1025
user_agent=<DICE 1.8.20.1>;reg-id=0
[root@proxy ~]# ngrep -W byline -d eth0 port 5060 interface: eth0 (178.79.X.X/255.255.255.0) filter: (ip or ip6) and ( port 5060 ) # T 94.94.X.X:49519 -> 178.79.x.x:5060 [AP] REGISTER sip:test.domain;transport=tcp;hide SIP/2.0. Via: SIP/2.0/TCP 94.94.X.X:49519;rport;branch=z9hG4bKPje03c11fd-d742-4502-98d8-69ca456ddd56;alias. Max-Forwards: 70. From: sip:528@test.domain;hide;tag=64c9e9b9-cb23-4a79-bb0d-2497135e449b. To: sip:528@test.domain;hide. Call-ID: 3ebadaeb-e6d6-42e3-9362-5ca4b5ddd8e1. CSeq: 30034 REGISTER. User-Agent: PJSUA v2.1-svn Linux-3.10.5.201/x86_64/glibc-2.17. Contact: sip:528@94.94.X.X:1274;transport=TCP;ob. Expires: 300. Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS. Content-Length: 0. .
# T 178.79.x.x:5060 -> 94.94.X.X:49519 [AP] SIP/2.0 401 Unauthorized. Via: SIP/2.0/TCP 94.94.X.X:49519;rport=49519;branch=z9hG4bKPje03c11fd-d742-4502-98d8-69ca456ddd56;alias. From: sip:528@test.domain;hide;tag=64c9e9b9-cb23-4a79-bb0d-2497135e449b. To: sip:528@test.domain;hide;tag=333a0370df4a40d5d5a0c21bb156e2a6.9167. Call-ID: 3ebadaeb-e6d6-42e3-9362-5ca4b5ddd8e1. CSeq: 30034 REGISTER. WWW-Authenticate: Digest realm="test.domain", nonce="Ugo+VFIKPSgVy8fh9fSy9SDLvT0wO4QV". Server: kamailio (4.0.2 (x86_64/linux)). Content-Length: 0. .
# U 178.79.x.x:5060 -> 94.94.X.X:1025 .... ## T 94.94.X.X:49519 -> 178.79.x.x:5060 [AP] REGISTER sip:test.domain;transport=tcp;hide SIP/2.0. Via: SIP/2.0/TCP 94.94.X.X:49519;rport;branch=z9hG4bKPj68a9c487-ae8c-428a-8c50-dd195a46a6bf;alias. Max-Forwards: 70. From: sip:528@test.domain;hide;tag=64c9e9b9-cb23-4a79-bb0d-2497135e449b. To: sip:528@test.domain;hide. Call-ID: 3ebadaeb-e6d6-42e3-9362-5ca4b5ddd8e1. CSeq: 30035 REGISTER. User-Agent: PJSUA v2.1-svn Linux-3.10.5.201/x86_64/glibc-2.17. Contact: sip:528@94.94.X.X:1274;transport=TCP;ob. Expires: 300. Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS. Authorization: Digest username="528", realm="test.domain", nonce="Ugo+VFIKPSgVy8fh9fSy9SDLvT0wO4QV", uri="sip:test.domain;transport=tcp;hide", response="ac1c7311ccb887fc8fb494d8ebf1bd36". Content-Length: 0. .
# T 178.79.x.x:5060 -> 94.94.X.X:49519 [AP] SIP/2.0 200 OK. Via: SIP/2.0/TCP 94.94.X.X:49519;rport=49519;branch=z9hG4bKPj68a9c487-ae8c-428a-8c50-dd195a46a6bf;alias. From: sip:528@test.domain;hide;tag=64c9e9b9-cb23-4a79-bb0d-2497135e449b. To: sip:528@test.domain;hide;tag=333a0370df4a40d5d5a0c21bb156e2a6.5440. Call-ID: 3ebadaeb-e6d6-42e3-9362-5ca4b5ddd8e1. CSeq: 30035 REGISTER. Contact: sip:528@94.94.X.X:1274;transport=TCP;ob;expires=300. Server: kamailio (4.0.2 (x86_64/linux)). Content-Length: 0. .
## U 94.94.X.X:1025 -> 178.79.x.x:5060 INVITE sip:528@test.domain:5060 SIP/2.0. Via: SIP/2.0/UDP 94.94.X.X:5060;branch=z9hG4bK69275d13;rport. Max-Forwards: 70. From: "asterisk" sip:512@test.domain;tag=as4531a4ee. To: sip:528@test.domain:5060. Contact: sip:512@94.94.X.X:5060. Call-ID: 2ea7e0f3173949d40b69385b62b79d6a@test.domain. CSeq: 102 INVITE. User-Agent: DICE 1.8.20.1. Date: Tue, 13 Aug 2013 14:04:11 GMT. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH. Supported: replaces, timer. Content-Type: application/sdp. Content-Length: 262. . v=0. o=root 883039875 883039875 IN IP4 94.94.X.X. s=Asterisk PBX 11.3.0. c=IN IP4 94.94.X.X. t=0 0. m=audio 18120 RTP/AVP 0 110 101. a=rtpmap:0 PCMU/8000. a=rtpmap:110 speex/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=sendrecv.
# U 178.79.x.x:5060 -> 94.94.X.X:1025 SIP/2.0 407 Proxy Authentication Required. Via: SIP/2.0/UDP 94.94.X.X:5060;branch=z9hG4bK69275d13;rport=1025. From: "asterisk" sip:512@test.domain;tag=as4531a4ee. To: sip:528@test.domain:5060;tag=333a0370df4a40d5d5a0c21bb156e2a6.038f. Call-ID: 2ea7e0f3173949d40b69385b62b79d6a@test.domain. CSeq: 102 INVITE. Proxy-Authenticate: Digest realm="test.domain", nonce="Ugo+XVIKPTFBd9tXC0QemfR9bHiZXO6x". Server: kamailio (4.0.2 (x86_64/linux)). Content-Length: 0. .
# U 94.94.X.X:1025 -> 178.79.x.x:5060 INVITE sip:528@test.domain:5060 SIP/2.0. Via: SIP/2.0/UDP 94.94.X.X:5060;branch=z9hG4bK69275d13;rport. Max-Forwards: 70. From: "asterisk" sip:512@test.domain;tag=as4531a4ee. To: sip:528@test.domain:5060. Contact: sip:512@94.94.X.X:5060. Call-ID: 2ea7e0f3173949d40b69385b62b79d6a@test.domain. CSeq: 102 INVITE. User-Agent: DICE 1.8.20.1. Date: Tue, 13 Aug 2013 14:04:11 GMT. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH. Supported: replaces, timer. Content-Type: application/sdp. Content-Length: 262. . v=0. o=root 883039875 883039875 IN IP4 94.94.X.X. s=Asterisk PBX 11.3.0. c=IN IP4 94.94.X.X. t=0 0. m=audio 18120 RTP/AVP 0 110 101. a=rtpmap:0 PCMU/8000. a=rtpmap:110 speex/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=sendrecv.
# U 178.79.x.x:5060 -> 94.94.X.X:1025 SIP/2.0 407 Proxy Authentication Required. Via: SIP/2.0/UDP 94.94.X.X:5060;branch=z9hG4bK69275d13;rport=1025. From: "asterisk" sip:512@test.domain;tag=as4531a4ee. To: sip:528@test.domain:5060;tag=333a0370df4a40d5d5a0c21bb156e2a6.038f. Call-ID: 2ea7e0f3173949d40b69385b62b79d6a@test.domain. CSeq: 102 INVITE. Proxy-Authenticate: Digest realm="test.domain", nonce="Ugo+XVIKPTFBd9tXC0QemfR9bHiZXO6x". Server: kamailio (4.0.2 (x86_64/linux)). Content-Length: 0. .
# U 94.94.X.X:1025 -> 178.79.x.x:5060 ACK sip:528@test.domain:5060 SIP/2.0. Via: SIP/2.0/UDP 94.94.X.X:5060;branch=z9hG4bK69275d13;rport. Max-Forwards: 70. From: "asterisk" sip:512@test.domain;tag=as4531a4ee. To: sip:528@test.domain:5060;tag=333a0370df4a40d5d5a0c21bb156e2a6.038f. Contact: sip:512@94.94.X.X:5060. Call-ID: 2ea7e0f3173949d40b69385b62b79d6a@test.domain. CSeq: 102 ACK. User-Agent: DICE 1.8.20.1. Content-Length: 0. .
# U 94.94.X.X:1025 -> 178.79.x.x:5060 ACK sip:528@test.domain:5060 SIP/2.0. Via: SIP/2.0/UDP 94.94.X.X:5060;branch=z9hG4bK69275d13;rport. Max-Forwards: 70. From: "asterisk" sip:512@test.domain;tag=as4531a4ee. To: sip:528@test.domain:5060. Contact: sip:512@94.94.X.X:5060. Call-ID: 2ea7e0f3173949d40b69385b62b79d6a@test.domain. CSeq: 102 ACK. User-Agent: DICE 1.8.20.1. Content-Length: 0. .
------------------ This is the client behind NAT ---------------------------
REGISTER sip:test.domain;transport=tcp;hide SIP/2.0 Via: SIP/2.0/TCP 192.168.2.90:49519;rport;branch=z9hG4bKPj92d04d45-bd84-45b3-9439-563ebfaebf00;alias Max-Forwards: 70 From: sip:528@test.domain;hide;tag=ab763476-0fdb-4d85-9064-1deabcaa23e4 To: sip:528@test.domain;hide Call-ID: 3ebadaeb-e6d6-42e3-9362-5ca4b5ddd8e1 CSeq: 30028 REGISTER User-Agent: PJSUA v2.1-svn Linux-3.10.5.201/x86_64/glibc-2.17 Contact: sip:528@192.168.2.90:5060;transport=TCP;ob Expires: 300 Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Content-Length: 0
--end msg-- 15:54:02.580 pjsua_acc.c .Acc 0: Registration sent
15:54:02.681 tcpc0x1515a08 !TCP transport 192.168.2.90:49519 is connected to 178.79.x.x:5060
15:54:02.681 pjsua_app.c SIP TCP transport is connected to [178.79.x.x:5060] 15:54:02.785 pjsua_core.c .RX 538 bytes Response msg 401/REGISTER/cseq=30028 (rdata0x1515cf8) from TCP 178.79.x.x:5060: SIP/2.0 401 Unauthorized Via: SIP/2.0/TCP 192.168.2.90:49519;rport=49519;branch=z9hG4bKPj92d04d45-bd84-45b3-9439-563ebfaebf00;alias From: sip:528@test.domain;hide;tag=ab763476-0fdb-4d85-9064-1deabcaa23e4 To: sip:528@test.domain;hide;tag=333a0370df4a40d5d5a0c21bb156e2a6.9663 Call-ID: 3ebadaeb-e6d6-42e3-9362-5ca4b5ddd8e1 CSeq: 30028 REGISTER WWW-Authenticate: Digest realm="test.domain", nonce="Ugo7/FIKOtBneWAQzD/VoJuucFT3cNuF" Server: kamailio (4.0.2 (x86_64/linux)) Content-Length: 0
--end msg-- 15:54:02.785 pjsua_core.c ....TX 849 bytes Request msg REGISTER/cseq=30029 (tdta0x1513000) to TCP 178.79.x.x:5060: REGISTER sip:test.domain;transport=tcp;hide SIP/2.0 Via: SIP/2.0/TCP 192.168.2.90:49519;rport;branch=z9hG4bKPj03a4a7eb-d086-4add-b84c-ca5c6cf4d05d;alias Max-Forwards: 70 From: sip:528@test.domain;hide;tag=ab763476-0fdb-4d85-9064-1deabcaa23e4 To: sip:528@test.domain;hide Call-ID: 3ebadaeb-e6d6-42e3-9362-5ca4b5ddd8e1 CSeq: 30029 REGISTER User-Agent: PJSUA v2.1-svn Linux-3.10.5.201/x86_64/glibc-2.17 Contact: sip:528@192.168.2.90:5060;transport=TCP;ob Expires: 300 Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Authorization: Digest username="528", realm="test.domain", nonce="Ugo7/FIKOtBneWAQzD/VoJuucFT3cNuF", uri="sip:test.domain;transport=tcp;hide", response="c3dd687602ec35b2403b3eb142b496f5" Content-Length: 0
--end msg-- 15:54:02.890 pjsua_core.c .RX 495 bytes Response msg 200/REGISTER/cseq=30029 (rdata0x1515cf8) from TCP 178.79.x.x:5060: SIP/2.0 200 OK Via: SIP/2.0/TCP 192.168.2.90:49519;rport=49519;branch=z9hG4bKPj03a4a7eb-d086-4add-b84c-ca5c6cf4d05d;alias From: sip:528@test.domain;hide;tag=ab763476-0fdb-4d85-9064-1deabcaa23e4 To: sip:528@test.domain;hide;tag=333a0370df4a40d5d5a0c21bb156e2a6.189b Call-ID: 3ebadaeb-e6d6-42e3-9362-5ca4b5ddd8e1 CSeq: 30029 REGISTER Contact: sip:528@192.168.2.90:5060;transport=TCP;ob;expires=300 Server: kamailio (4.0.2 (x86_64/linux)) Content-Length: 0
--end msg-- 15:54:02.890 pjsua_acc.c ....SIP outbound status for acc 0 is not active 15:54:02.890 pjsua_acc.c ....sip:528@test.domain;transport=tcp;hide: registration success, status=200 (OK), will re-register in 300 seconds
Daniel
On 8/13/13 3:23 PM, Roberto Fichera wrote:
On 08/13/2013 03:15 PM, Roberto Fichera wrote:
On 08/13/2013 02:33 PM, Daniel-Constantin Mierla wrote: > Hello, > > On 8/13/13 1:10 PM, Roberto Fichera wrote: >> On 08/13/2013 12:03 PM, Daniel-Constantin Mierla wrote: >>> Hello, >>> >>> you should grab the ngrep for such call to understand better what happens. Also, dumping the location records >>> will be >>> useful (kamctl ul show). >>> >>> Also, be sure that tcp connection lifetime is long enough to survive re-registration. To avoid trying to open >>> connections behind nat, use set_forward_no_connect() for calls involving nat traversal. >> I'm using the default conf coming from fedora rpm. So, mainly the problem seems related to kamailio >> which doesn't reuse the TCP port used by NATed clients. I've also notice that the received >> field isn't set at all, so this means that the contact will not get aliased at all. >> >> I would really like to have a look to a working cfg file for TCP NATed clients that reuse the TCP port. >> Even better if the configuration is based on the fedora default rpm. > if received is not set, then means the register was not detected as coming from behind nat. Is the phone using > stun? I'm testing with a normal rtpproxy configuration. BTW udp -> udp work perfectly.
> Again, put here the ngrep for registration and a call to see if something is wrong with signaling. There is no help > that we can provide otherwise. The default config works fine for tcp and natted clients, I use it everywhere for > this > case without issues. I tried the default cfg enabling both NAT and RTPproxy, but seems that kamailio doesn't reuse TCP ports. Anyway, this is a call from UDP (512) -> TCP (526) both behind the same NAT, from kamailio point of view
I forgot to say that the received field is now present because I've changed the route[NATDETECT] in the default configuration as
route[NATDETECT] { #!ifdef WITH_NAT force_rport();
-->>> if (nat_uac_test("19") || proto != UDP) { if (is_method("REGISTER")) { fix_nated_register(); } else { fix_nated_contact(); } setflag(FLT_NATS); } #!endif return; }
[root@proxy ~]# kamctl ul show 526 Contact:: sip:526@94.94.X.X:1238;transport=TCP;ob;q=;expires=537;flags=0x0;cflags=0x40;state=0;socket=tcp:178.79.x.x:5060;methods=0x1FDF;received=sip:94.94.X.X:61922;transport=TCP;user_agent=<DICE
Smartphone 1.0/iPhone>;reg-id=0 [root@proxy ~]# kamctl ul show 512 Contact:: sip:512@94.94.X.X:5060;q=;expires=32;flags=0x0;cflags=0x40;state=0;socket=udp:178.79.x.x:5060;methods=0xFFFFFFFF;received=sip:94.94.X.X:1025;user_agent=<DICE
1.8.20.1>;reg-id=0 [root@proxy ~]#
U 94.94.X.X:1025 -> 178.79.x.x:5060 INVITE sip:526@test.domain:5060 SIP/2.0. Via: SIP/2.0/UDP 94.94.X.X:5060;branch=z9hG4bK4a420cee;rport. Max-Forwards: 70. From: "asterisk" sip:512@test.domain;tag=as76007db0. To: sip:526@test.domain:5060. Contact: sip:512@94.94.X.X:5060. Call-ID: 068a5a23639785a7583d952d6f9bca84@test.domain. CSeq: 102 INVITE. User-Agent: DICE 1.8.20.1. Date: Tue, 13 Aug 2013 13:04:30 GMT. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH. Supported: replaces, timer. Content-Type: application/sdp. Content-Length: 264. . v=0. o=root 1263161426 1263161426 IN IP4 94.94.X.X. s=Asterisk PBX 11.3.0. c=IN IP4 94.94.X.X. t=0 0. m=audio 10782 RTP/AVP 0 110 101. a=rtpmap:0 PCMU/8000. a=rtpmap:110 speex/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=sendrecv.
# U 178.79.x.x:5060 -> 94.94.X.X:1025 SIP/2.0 407 Proxy Authentication Required. Via: SIP/2.0/UDP 94.94.X.X:5060;branch=z9hG4bK4a420cee;rport=1025. From: "asterisk" sip:512@test.domain;tag=as76007db0. To: sip:526@test.domain:5060;tag=333a0370df4a40d5d5a0c21bb156e2a6.4a00. Call-ID: 068a5a23639785a7583d952d6f9bca84@test.domain. CSeq: 102 INVITE. Proxy-Authenticate: Digest realm="test.domain", nonce="UgowYFIKLzQXDSUWUpy4xlhdXZ7WNjPt". Server: kamailio (4.0.2 (x86_64/linux)). Content-Length: 0. .
# U 94.94.X.X:1025 -> 178.79.x.x:5060 INVITE sip:526@test.domain:5060 SIP/2.0. Via: SIP/2.0/UDP 94.94.X.X:5060;branch=z9hG4bK4a420cee;rport. Max-Forwards: 70. From: "asterisk" sip:512@test.domain;tag=as76007db0. To: sip:526@test.domain:5060. Contact: sip:512@94.94.X.X:5060. Call-ID: 068a5a23639785a7583d952d6f9bca84@test.domain. CSeq: 102 INVITE. User-Agent: DICE 1.8.20.1. Date: Tue, 13 Aug 2013 13:04:30 GMT. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH. Supported: replaces, timer. Content-Type: application/sdp. Content-Length: 264. . v=0. o=root 1263161426 1263161426 IN IP4 94.94.X.X. s=Asterisk PBX 11.3.0. c=IN IP4 94.94.X.X. t=0 0. m=audio 10782 RTP/AVP 0 110 101. a=rtpmap:0 PCMU/8000. a=rtpmap:110 speex/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=sendrecv.
# U 178.79.x.x:5060 -> 94.94.X.X:1025 SIP/2.0 407 Proxy Authentication Required. Via: SIP/2.0/UDP 94.94.X.X:5060;branch=z9hG4bK4a420cee;rport=1025. From: "asterisk" sip:512@test.domain;tag=as76007db0. To: sip:526@test.domain:5060;tag=333a0370df4a40d5d5a0c21bb156e2a6.4a00. Call-ID: 068a5a23639785a7583d952d6f9bca84@test.domain. CSeq: 102 INVITE. Proxy-Authenticate: Digest realm="test.domain", nonce="UgowYFIKLzQXDSUWUpy4xlhdXZ7WNjPt". Server: kamailio (4.0.2 (x86_64/linux)). Content-Length: 0. .
# U 94.94.X.X:1025 -> 178.79.x.x:5060 ACK sip:526@test.domain:5060 SIP/2.0. Via: SIP/2.0/UDP 94.94.X.X:5060;branch=z9hG4bK4a420cee;rport. Max-Forwards: 70. From: "asterisk" sip:512@test.domain;tag=as76007db0. To: sip:526@test.domain:5060;tag=333a0370df4a40d5d5a0c21bb156e2a6.4a00. Contact: sip:512@94.94.X.X:5060. Call-ID: 068a5a23639785a7583d952d6f9bca84@test.domain. CSeq: 102 ACK. User-Agent: DICE 1.8.20.1. Content-Length: 0. .
# U 94.94.X.X:1025 -> 178.79.x.x:5060 ACK sip:526@test.domain:5060 SIP/2.0. Via: SIP/2.0/UDP 94.94.X.X:5060;branch=z9hG4bK4a420cee;rport. Max-Forwards: 70. From: "asterisk" sip:512@test.domain;tag=as76007db0. To: sip:526@test.domain:5060. Contact: sip:512@94.94.X.X:5060. Call-ID: 068a5a23639785a7583d952d6f9bca84@test.domain. CSeq: 102 ACK. User-Agent: DICE 1.8.20.1. Content-Length: 0. .
# T 178.79.x.x:5060 -> 94.94.X.X:61922 [AP] ACK sip:526@94.94.X.X:1238;transport=TCP;ob SIP/2.0. Via: SIP/2.0/TCP 178.79.x.x;branch=z9hG4bKcydzigwkX. Via: SIP/2.0/UDP 94.94.X.X:5060;branch=z9hG4bK4a420cee;rport=1025. Max-Forwards: 16. From: "asterisk" sip:512@test.domain;tag=as76007db0. To: sip:526@test.domain:5060. Contact: sip:512@94.94.X.X:1025. Call-ID: 068a5a23639785a7583d952d6f9bca84@test.domain. CSeq: 102 ACK. User-Agent: DICE 1.8.20.1. Content-Length: 0. .
# T 178.79.x.x:5060 -> 94.94.X.X:61922 [AP] ACK sip:526@94.94.X.X:1238;transport=TCP;ob SIP/2.0. Via: SIP/2.0/TCP 178.79.x.x;branch=z9hG4bKcydzigwkX. Via: SIP/2.0/UDP 94.94.X.X:5060;branch=z9hG4bK4a420cee;rport=1025. Max-Forwards: 16. From: "asterisk" sip:512@test.domain;tag=as76007db0. To: sip:526@test.domain:5060. Contact: sip:512@94.94.X.X:1025. Call-ID: 068a5a23639785a7583d952d6f9bca84@test.domain. CSeq: 102 ACK. User-Agent: DICE 1.8.20.1. Content-Length: 0. .
> Cheers, > Daniel >> Cheers, >> Roberto Fichera. >> >>> Cheers, >>> Daniel >>> >>> On 7/30/13 6:44 PM, Roberto Fichera wrote: >>>> Hi All, >>>> >>>> Sorry for cross-posting this email to PJLIB, but maybe there are some things related. >>>> Anyhow! I'm having problems on kamailio v4.0.2 under Fedora 18 64bit and TCP client like iPhone using PJSIP >>>> as SIP >>>> library. >>>> Basically once the iPhone side in close the call (TCP->UDP) I'm getting the error below. Kamailio is running >>>> under >>>> a VPS >>>> without >>>> NATed network so it uses a real public address. Furthermore, note that tcp_main is answering to a >>>> 192.168.2.98 ip >>>> address >>>> which is the iPhone client. This looks really strange to me since it should answer directly to the >>>> public/port used >>>> for >>>> the registration >>>> and not to a such kind of reserved address. The kamilio configuration is basically the default with a very few >>>> changes >>>> like NAT, rtpproxy and postgresql backend. >>>> >>>> This problems doesn't happen at all when using UDP->UDP calls. But I cannot use it because as you certain >>>> know UDP >>>> connection under iPhone will not work when the application run in background mode. >>>> >>>> Can someone suggest how to solve this issue or maybe suggest a TCP working solution for iPhone? >>>> >>>> Thanks in advance. >>>> Roberto Fichera. >>>> >>>> Jul 30 16:21:53 proxy /usr/sbin/kamailio[9502]: ERROR: <core> [tcp_main.c:4432]: tcpconn_main_timeout(): connect >>>> 192.168.2.98:5060 failed (timeout) >>>> Jul 30 16:21:55 proxy rtpproxy[2262]: DBUG:get_command: received command "9483_9 D >>>> 12d1d19926c4ff742a52f0c855b1bb83@94.94.x.x:5060 as74e0c388 GROahimCK6KTrl5CkYEg7nuoPIIXZ8cj" >>>> Jul 30 16:21:55 proxy rtpproxy[2262]: INFO:handle_delete: forcefully deleting session 1 on ports 15604/17354 >>>> Jul 30 16:21:55 proxy rtpproxy[2262]: INFO:remove_session: RTP stats: 354 in from callee, 603 in from caller, >>>> 957 >>>> relayed, 0 dropped >>>> Jul 30 16:21:55 proxy rtpproxy[2262]: INFO:remove_session: RTCP stats: 5 in from callee, 2 in from caller, 7 >>>> relayed, 0 >>>> dropped >>>> Jul 30 16:21:55 proxy rtpproxy[2262]: INFO:remove_session: session on ports 15604/17354 is cleaned up >>>> Jul 30 16:21:55 proxy rtpproxy[2262]: DBUG:doreply: sending reply "9483_9 0 >>>> Jul 30 16:21:55 proxy rtpproxy[2262]: " >>>> Jul 30 16:22:04 proxy /usr/sbin/kamailio[9502]: ERROR: <core> [tcp_main.c:4432]: tcpconn_main_timeout(): connect >>>> 192.168.2.98:5060 failed (timeout) >>>> Jul 30 16:22:14 proxy rtpproxy[2262]: DBUG:get_command: received command "9496_16 D >>>> 12d1d19926c4ff742a52f0c855b1bb83@94.94.x.x:5060 GROahimCK6KTrl5CkYEg7nuoPIIXZ8cj as74e0c388" >>>> Jul 30 16:22:14 proxy rtpproxy[2262]: INFO:handle_command: delete request failed: session >>>> 12d1d19926c4ff742a52f0c855b1bb83@94.94.x.x:5060, tags GROahimCK6KTrl5CkYEg7nuoPIIXZ8cj/as74e0c388 not found >>>> Jul 30 16:22:14 proxy rtpproxy[2262]: DBUG:doreply: sending reply "9496_16 E8 >>>> Jul 30 16:22:14 proxy rtpproxy[2262]: " >>>> >>>> >>>> _______________________________________________ >>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >>>> sr-users@lists.sip-router.org >>>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Hello,
On 8/14/13 11:16 AM, Roberto Fichera wrote:
On 08/14/2013 10:19 AM, Daniel-Constantin Mierla wrote:
Hi,
Hello,
as you can see in the REGISTER, the phone give a public IP where it can be contacted and kamailio tries to deliver to that address sip:528@94.94.X.X:1274;transport=TCP
REGISTER comes from another port, but that is allowed in SIP.
Yep! That's the problem
You should disable stun in the client and let the server care of nat traversal.
As already said all client doesn't use any stun-
then how they discover the public IP address if they are behind NAT? Or did I misunderstand that they are behind NAT?
If they are on the public IP, they they should listen on the port advertised in the Contact header of registration and accept the connection from Kamailio.
Cheers, Daniel
Or you can try to add flag 128 to nat test:
Also, you can try to add an alias for contact port:
Ok! I'll try both!
Cheers, Roberto Fichera.
Cheers, Daniel
On 8/13/13 4:17 PM, Roberto Fichera wrote:
On 08/13/2013 03:32 PM, Roberto Fichera wrote:
On 08/13/2013 03:25 PM, Daniel-Constantin Mierla wrote:
Can you get a ngrep trace for a registration as well (for the phone using tcp)?
Ok! I'll use pjsua from my local machine connecting in the same way as the TCP client was doing. The TCP client it's an iPhone using the same pjlib library.
I can confirm that the default cfg for TCP client doesn't work for me. My cfg file is attached. The TCP client doesn't receive any package at INVITE. Finally in /var/log/message I get this log below:
Aug 13 14:05:37 proxy /usr/sbin/kamailio[8401]: ERROR: <core> [tcp_main.c:4247]: handle_tcpconn_ev(): connect 94.94.X.X:1274 failed
Contact:: sip:528@94.94.X.X:1274;transport=TCP;ob;q=;expires=38;flags=0x0;cflags=0x0;state=0;socket=tcp:178.79.X.X:5060;methods=0x1FDF;user_agent=<PJSUA
v2.1-svn Linux-3.10.5.201/x86_64/glibc-2.17>;reg-id=0 [root@proxy ~]# kamctl ul show 512 Contact:: sip:512@94.94.X.X:5060;q=;expires=61;flags=0x0;cflags=0x40;state=0;socket=udp:178.79.X.X:5060;methods=0xFFFFFFFF;received=sip:94.94.X.X:1025
user_agent=<DICE 1.8.20.1>;reg-id=0
[root@proxy ~]# ngrep -W byline -d eth0 port 5060 interface: eth0 (178.79.X.X/255.255.255.0) filter: (ip or ip6) and ( port 5060 ) # T 94.94.X.X:49519 -> 178.79.x.x:5060 [AP] REGISTER sip:test.domain;transport=tcp;hide SIP/2.0. Via: SIP/2.0/TCP 94.94.X.X:49519;rport;branch=z9hG4bKPje03c11fd-d742-4502-98d8-69ca456ddd56;alias. Max-Forwards: 70. From: sip:528@test.domain;hide;tag=64c9e9b9-cb23-4a79-bb0d-2497135e449b. To: sip:528@test.domain;hide. Call-ID: 3ebadaeb-e6d6-42e3-9362-5ca4b5ddd8e1. CSeq: 30034 REGISTER. User-Agent: PJSUA v2.1-svn Linux-3.10.5.201/x86_64/glibc-2.17. Contact: sip:528@94.94.X.X:1274;transport=TCP;ob. Expires: 300. Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS. Content-Length: 0. .
# T 178.79.x.x:5060 -> 94.94.X.X:49519 [AP] SIP/2.0 401 Unauthorized. Via: SIP/2.0/TCP 94.94.X.X:49519;rport=49519;branch=z9hG4bKPje03c11fd-d742-4502-98d8-69ca456ddd56;alias. From: sip:528@test.domain;hide;tag=64c9e9b9-cb23-4a79-bb0d-2497135e449b. To: sip:528@test.domain;hide;tag=333a0370df4a40d5d5a0c21bb156e2a6.9167. Call-ID: 3ebadaeb-e6d6-42e3-9362-5ca4b5ddd8e1. CSeq: 30034 REGISTER. WWW-Authenticate: Digest realm="test.domain", nonce="Ugo+VFIKPSgVy8fh9fSy9SDLvT0wO4QV". Server: kamailio (4.0.2 (x86_64/linux)). Content-Length: 0. .
# U 178.79.x.x:5060 -> 94.94.X.X:1025 .... ## T 94.94.X.X:49519 -> 178.79.x.x:5060 [AP] REGISTER sip:test.domain;transport=tcp;hide SIP/2.0. Via: SIP/2.0/TCP 94.94.X.X:49519;rport;branch=z9hG4bKPj68a9c487-ae8c-428a-8c50-dd195a46a6bf;alias. Max-Forwards: 70. From: sip:528@test.domain;hide;tag=64c9e9b9-cb23-4a79-bb0d-2497135e449b. To: sip:528@test.domain;hide. Call-ID: 3ebadaeb-e6d6-42e3-9362-5ca4b5ddd8e1. CSeq: 30035 REGISTER. User-Agent: PJSUA v2.1-svn Linux-3.10.5.201/x86_64/glibc-2.17. Contact: sip:528@94.94.X.X:1274;transport=TCP;ob. Expires: 300. Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS. Authorization: Digest username="528", realm="test.domain", nonce="Ugo+VFIKPSgVy8fh9fSy9SDLvT0wO4QV", uri="sip:test.domain;transport=tcp;hide", response="ac1c7311ccb887fc8fb494d8ebf1bd36". Content-Length: 0. .
# T 178.79.x.x:5060 -> 94.94.X.X:49519 [AP] SIP/2.0 200 OK. Via: SIP/2.0/TCP 94.94.X.X:49519;rport=49519;branch=z9hG4bKPj68a9c487-ae8c-428a-8c50-dd195a46a6bf;alias. From: sip:528@test.domain;hide;tag=64c9e9b9-cb23-4a79-bb0d-2497135e449b. To: sip:528@test.domain;hide;tag=333a0370df4a40d5d5a0c21bb156e2a6.5440. Call-ID: 3ebadaeb-e6d6-42e3-9362-5ca4b5ddd8e1. CSeq: 30035 REGISTER. Contact: sip:528@94.94.X.X:1274;transport=TCP;ob;expires=300. Server: kamailio (4.0.2 (x86_64/linux)). Content-Length: 0. .
## U 94.94.X.X:1025 -> 178.79.x.x:5060 INVITE sip:528@test.domain:5060 SIP/2.0. Via: SIP/2.0/UDP 94.94.X.X:5060;branch=z9hG4bK69275d13;rport. Max-Forwards: 70. From: "asterisk" sip:512@test.domain;tag=as4531a4ee. To: sip:528@test.domain:5060. Contact: sip:512@94.94.X.X:5060. Call-ID: 2ea7e0f3173949d40b69385b62b79d6a@test.domain. CSeq: 102 INVITE. User-Agent: DICE 1.8.20.1. Date: Tue, 13 Aug 2013 14:04:11 GMT. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH. Supported: replaces, timer. Content-Type: application/sdp. Content-Length: 262. . v=0. o=root 883039875 883039875 IN IP4 94.94.X.X. s=Asterisk PBX 11.3.0. c=IN IP4 94.94.X.X. t=0 0. m=audio 18120 RTP/AVP 0 110 101. a=rtpmap:0 PCMU/8000. a=rtpmap:110 speex/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=sendrecv.
# U 178.79.x.x:5060 -> 94.94.X.X:1025 SIP/2.0 407 Proxy Authentication Required. Via: SIP/2.0/UDP 94.94.X.X:5060;branch=z9hG4bK69275d13;rport=1025. From: "asterisk" sip:512@test.domain;tag=as4531a4ee. To: sip:528@test.domain:5060;tag=333a0370df4a40d5d5a0c21bb156e2a6.038f. Call-ID: 2ea7e0f3173949d40b69385b62b79d6a@test.domain. CSeq: 102 INVITE. Proxy-Authenticate: Digest realm="test.domain", nonce="Ugo+XVIKPTFBd9tXC0QemfR9bHiZXO6x". Server: kamailio (4.0.2 (x86_64/linux)). Content-Length: 0. .
# U 94.94.X.X:1025 -> 178.79.x.x:5060 INVITE sip:528@test.domain:5060 SIP/2.0. Via: SIP/2.0/UDP 94.94.X.X:5060;branch=z9hG4bK69275d13;rport. Max-Forwards: 70. From: "asterisk" sip:512@test.domain;tag=as4531a4ee. To: sip:528@test.domain:5060. Contact: sip:512@94.94.X.X:5060. Call-ID: 2ea7e0f3173949d40b69385b62b79d6a@test.domain. CSeq: 102 INVITE. User-Agent: DICE 1.8.20.1. Date: Tue, 13 Aug 2013 14:04:11 GMT. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH. Supported: replaces, timer. Content-Type: application/sdp. Content-Length: 262. . v=0. o=root 883039875 883039875 IN IP4 94.94.X.X. s=Asterisk PBX 11.3.0. c=IN IP4 94.94.X.X. t=0 0. m=audio 18120 RTP/AVP 0 110 101. a=rtpmap:0 PCMU/8000. a=rtpmap:110 speex/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=sendrecv.
# U 178.79.x.x:5060 -> 94.94.X.X:1025 SIP/2.0 407 Proxy Authentication Required. Via: SIP/2.0/UDP 94.94.X.X:5060;branch=z9hG4bK69275d13;rport=1025. From: "asterisk" sip:512@test.domain;tag=as4531a4ee. To: sip:528@test.domain:5060;tag=333a0370df4a40d5d5a0c21bb156e2a6.038f. Call-ID: 2ea7e0f3173949d40b69385b62b79d6a@test.domain. CSeq: 102 INVITE. Proxy-Authenticate: Digest realm="test.domain", nonce="Ugo+XVIKPTFBd9tXC0QemfR9bHiZXO6x". Server: kamailio (4.0.2 (x86_64/linux)). Content-Length: 0. .
# U 94.94.X.X:1025 -> 178.79.x.x:5060 ACK sip:528@test.domain:5060 SIP/2.0. Via: SIP/2.0/UDP 94.94.X.X:5060;branch=z9hG4bK69275d13;rport. Max-Forwards: 70. From: "asterisk" sip:512@test.domain;tag=as4531a4ee. To: sip:528@test.domain:5060;tag=333a0370df4a40d5d5a0c21bb156e2a6.038f. Contact: sip:512@94.94.X.X:5060. Call-ID: 2ea7e0f3173949d40b69385b62b79d6a@test.domain. CSeq: 102 ACK. User-Agent: DICE 1.8.20.1. Content-Length: 0. .
# U 94.94.X.X:1025 -> 178.79.x.x:5060 ACK sip:528@test.domain:5060 SIP/2.0. Via: SIP/2.0/UDP 94.94.X.X:5060;branch=z9hG4bK69275d13;rport. Max-Forwards: 70. From: "asterisk" sip:512@test.domain;tag=as4531a4ee. To: sip:528@test.domain:5060. Contact: sip:512@94.94.X.X:5060. Call-ID: 2ea7e0f3173949d40b69385b62b79d6a@test.domain. CSeq: 102 ACK. User-Agent: DICE 1.8.20.1. Content-Length: 0. .
------------------ This is the client behind NAT ---------------------------
REGISTER sip:test.domain;transport=tcp;hide SIP/2.0 Via: SIP/2.0/TCP 192.168.2.90:49519;rport;branch=z9hG4bKPj92d04d45-bd84-45b3-9439-563ebfaebf00;alias Max-Forwards: 70 From: sip:528@test.domain;hide;tag=ab763476-0fdb-4d85-9064-1deabcaa23e4 To: sip:528@test.domain;hide Call-ID: 3ebadaeb-e6d6-42e3-9362-5ca4b5ddd8e1 CSeq: 30028 REGISTER User-Agent: PJSUA v2.1-svn Linux-3.10.5.201/x86_64/glibc-2.17 Contact: sip:528@192.168.2.90:5060;transport=TCP;ob Expires: 300 Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Content-Length: 0
--end msg-- 15:54:02.580 pjsua_acc.c .Acc 0: Registration sent
15:54:02.681 tcpc0x1515a08 !TCP transport 192.168.2.90:49519 is connected to 178.79.x.x:5060
15:54:02.681 pjsua_app.c SIP TCP transport is connected to [178.79.x.x:5060] 15:54:02.785 pjsua_core.c .RX 538 bytes Response msg 401/REGISTER/cseq=30028 (rdata0x1515cf8) from TCP 178.79.x.x:5060: SIP/2.0 401 Unauthorized Via: SIP/2.0/TCP 192.168.2.90:49519;rport=49519;branch=z9hG4bKPj92d04d45-bd84-45b3-9439-563ebfaebf00;alias From: sip:528@test.domain;hide;tag=ab763476-0fdb-4d85-9064-1deabcaa23e4 To: sip:528@test.domain;hide;tag=333a0370df4a40d5d5a0c21bb156e2a6.9663 Call-ID: 3ebadaeb-e6d6-42e3-9362-5ca4b5ddd8e1 CSeq: 30028 REGISTER WWW-Authenticate: Digest realm="test.domain", nonce="Ugo7/FIKOtBneWAQzD/VoJuucFT3cNuF" Server: kamailio (4.0.2 (x86_64/linux)) Content-Length: 0
--end msg-- 15:54:02.785 pjsua_core.c ....TX 849 bytes Request msg REGISTER/cseq=30029 (tdta0x1513000) to TCP 178.79.x.x:5060: REGISTER sip:test.domain;transport=tcp;hide SIP/2.0 Via: SIP/2.0/TCP 192.168.2.90:49519;rport;branch=z9hG4bKPj03a4a7eb-d086-4add-b84c-ca5c6cf4d05d;alias Max-Forwards: 70 From: sip:528@test.domain;hide;tag=ab763476-0fdb-4d85-9064-1deabcaa23e4 To: sip:528@test.domain;hide Call-ID: 3ebadaeb-e6d6-42e3-9362-5ca4b5ddd8e1 CSeq: 30029 REGISTER User-Agent: PJSUA v2.1-svn Linux-3.10.5.201/x86_64/glibc-2.17 Contact: sip:528@192.168.2.90:5060;transport=TCP;ob Expires: 300 Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Authorization: Digest username="528", realm="test.domain", nonce="Ugo7/FIKOtBneWAQzD/VoJuucFT3cNuF", uri="sip:test.domain;transport=tcp;hide", response="c3dd687602ec35b2403b3eb142b496f5" Content-Length: 0
--end msg-- 15:54:02.890 pjsua_core.c .RX 495 bytes Response msg 200/REGISTER/cseq=30029 (rdata0x1515cf8) from TCP 178.79.x.x:5060: SIP/2.0 200 OK Via: SIP/2.0/TCP 192.168.2.90:49519;rport=49519;branch=z9hG4bKPj03a4a7eb-d086-4add-b84c-ca5c6cf4d05d;alias From: sip:528@test.domain;hide;tag=ab763476-0fdb-4d85-9064-1deabcaa23e4 To: sip:528@test.domain;hide;tag=333a0370df4a40d5d5a0c21bb156e2a6.189b Call-ID: 3ebadaeb-e6d6-42e3-9362-5ca4b5ddd8e1 CSeq: 30029 REGISTER Contact: sip:528@192.168.2.90:5060;transport=TCP;ob;expires=300 Server: kamailio (4.0.2 (x86_64/linux)) Content-Length: 0
--end msg-- 15:54:02.890 pjsua_acc.c ....SIP outbound status for acc 0 is not active 15:54:02.890 pjsua_acc.c ....sip:528@test.domain;transport=tcp;hide: registration success, status=200 (OK), will re-register in 300 seconds
Daniel
On 8/13/13 3:23 PM, Roberto Fichera wrote:
On 08/13/2013 03:15 PM, Roberto Fichera wrote: > On 08/13/2013 02:33 PM, Daniel-Constantin Mierla wrote: >> Hello, >> >> On 8/13/13 1:10 PM, Roberto Fichera wrote: >>> On 08/13/2013 12:03 PM, Daniel-Constantin Mierla wrote: >>>> Hello, >>>> >>>> you should grab the ngrep for such call to understand better what happens. Also, dumping the location records >>>> will be >>>> useful (kamctl ul show). >>>> >>>> Also, be sure that tcp connection lifetime is long enough to survive re-registration. To avoid trying to open >>>> connections behind nat, use set_forward_no_connect() for calls involving nat traversal. >>> I'm using the default conf coming from fedora rpm. So, mainly the problem seems related to kamailio >>> which doesn't reuse the TCP port used by NATed clients. I've also notice that the received >>> field isn't set at all, so this means that the contact will not get aliased at all. >>> >>> I would really like to have a look to a working cfg file for TCP NATed clients that reuse the TCP port. >>> Even better if the configuration is based on the fedora default rpm. >> if received is not set, then means the register was not detected as coming from behind nat. Is the phone using >> stun? > I'm testing with a normal rtpproxy configuration. BTW udp -> udp work perfectly. > >> Again, put here the ngrep for registration and a call to see if something is wrong with signaling. There is no help >> that we can provide otherwise. The default config works fine for tcp and natted clients, I use it everywhere for >> this >> case without issues. > I tried the default cfg enabling both NAT and RTPproxy, but seems that kamailio doesn't reuse TCP ports. > Anyway, this is a call from UDP (512) -> TCP (526) both behind the same NAT, from kamailio point of view I forgot to say that the received field is now present because I've changed the route[NATDETECT] in the default configuration as
route[NATDETECT] { #!ifdef WITH_NAT force_rport();
-->>> if (nat_uac_test("19") || proto != UDP) { if (is_method("REGISTER")) { fix_nated_register(); } else { fix_nated_contact(); } setflag(FLT_NATS); } #!endif return; }
> [root@proxy ~]# kamctl ul show 526 > Contact:: > sip:526@94.94.X.X:1238;transport=TCP;ob;q=;expires=537;flags=0x0;cflags=0x40;state=0;socket=tcp:178.79.x.x:5060;methods=0x1FDF;received=sip:94.94.X.X:61922;transport=TCP;user_agent=<DICE > > > Smartphone 1.0/iPhone>;reg-id=0 > [root@proxy ~]# kamctl ul show 512 > Contact:: > sip:512@94.94.X.X:5060;q=;expires=32;flags=0x0;cflags=0x40;state=0;socket=udp:178.79.x.x:5060;methods=0xFFFFFFFF;received=sip:94.94.X.X:1025;user_agent=<DICE > > > 1.8.20.1>;reg-id=0 > [root@proxy ~]# > > > U 94.94.X.X:1025 -> 178.79.x.x:5060 > INVITE sip:526@test.domain:5060 SIP/2.0. > Via: SIP/2.0/UDP 94.94.X.X:5060;branch=z9hG4bK4a420cee;rport. > Max-Forwards: 70. > From: "asterisk" sip:512@test.domain;tag=as76007db0. > To: sip:526@test.domain:5060. > Contact: sip:512@94.94.X.X:5060. > Call-ID: 068a5a23639785a7583d952d6f9bca84@test.domain. > CSeq: 102 INVITE. > User-Agent: DICE 1.8.20.1. > Date: Tue, 13 Aug 2013 13:04:30 GMT. > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH. > Supported: replaces, timer. > Content-Type: application/sdp. > Content-Length: 264. > . > v=0. > o=root 1263161426 1263161426 IN IP4 94.94.X.X. > s=Asterisk PBX 11.3.0. > c=IN IP4 94.94.X.X. > t=0 0. > m=audio 10782 RTP/AVP 0 110 101. > a=rtpmap:0 PCMU/8000. > a=rtpmap:110 speex/8000. > a=rtpmap:101 telephone-event/8000. > a=fmtp:101 0-16. > a=ptime:20. > a=sendrecv. > > # > U 178.79.x.x:5060 -> 94.94.X.X:1025 > SIP/2.0 407 Proxy Authentication Required. > Via: SIP/2.0/UDP 94.94.X.X:5060;branch=z9hG4bK4a420cee;rport=1025. > From: "asterisk" sip:512@test.domain;tag=as76007db0. > To: sip:526@test.domain:5060;tag=333a0370df4a40d5d5a0c21bb156e2a6.4a00. > Call-ID: 068a5a23639785a7583d952d6f9bca84@test.domain. > CSeq: 102 INVITE. > Proxy-Authenticate: Digest realm="test.domain", nonce="UgowYFIKLzQXDSUWUpy4xlhdXZ7WNjPt". > Server: kamailio (4.0.2 (x86_64/linux)). > Content-Length: 0. > . > > # > U 94.94.X.X:1025 -> 178.79.x.x:5060 > INVITE sip:526@test.domain:5060 SIP/2.0. > Via: SIP/2.0/UDP 94.94.X.X:5060;branch=z9hG4bK4a420cee;rport. > Max-Forwards: 70. > From: "asterisk" sip:512@test.domain;tag=as76007db0. > To: sip:526@test.domain:5060. > Contact: sip:512@94.94.X.X:5060. > Call-ID: 068a5a23639785a7583d952d6f9bca84@test.domain. > CSeq: 102 INVITE. > User-Agent: DICE 1.8.20.1. > Date: Tue, 13 Aug 2013 13:04:30 GMT. > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH. > Supported: replaces, timer. > Content-Type: application/sdp. > Content-Length: 264. > . > v=0. > o=root 1263161426 1263161426 IN IP4 94.94.X.X. > s=Asterisk PBX 11.3.0. > c=IN IP4 94.94.X.X. > t=0 0. > m=audio 10782 RTP/AVP 0 110 101. > a=rtpmap:0 PCMU/8000. > a=rtpmap:110 speex/8000. > a=rtpmap:101 telephone-event/8000. > a=fmtp:101 0-16. > a=ptime:20. > a=sendrecv. > > # > U 178.79.x.x:5060 -> 94.94.X.X:1025 > SIP/2.0 407 Proxy Authentication Required. > Via: SIP/2.0/UDP 94.94.X.X:5060;branch=z9hG4bK4a420cee;rport=1025. > From: "asterisk" sip:512@test.domain;tag=as76007db0. > To: sip:526@test.domain:5060;tag=333a0370df4a40d5d5a0c21bb156e2a6.4a00. > Call-ID: 068a5a23639785a7583d952d6f9bca84@test.domain. > CSeq: 102 INVITE. > Proxy-Authenticate: Digest realm="test.domain", nonce="UgowYFIKLzQXDSUWUpy4xlhdXZ7WNjPt". > Server: kamailio (4.0.2 (x86_64/linux)). > Content-Length: 0. > . > > # > U 94.94.X.X:1025 -> 178.79.x.x:5060 > ACK sip:526@test.domain:5060 SIP/2.0. > Via: SIP/2.0/UDP 94.94.X.X:5060;branch=z9hG4bK4a420cee;rport. > Max-Forwards: 70. > From: "asterisk" sip:512@test.domain;tag=as76007db0. > To: sip:526@test.domain:5060;tag=333a0370df4a40d5d5a0c21bb156e2a6.4a00. > Contact: sip:512@94.94.X.X:5060. > Call-ID: 068a5a23639785a7583d952d6f9bca84@test.domain. > CSeq: 102 ACK. > User-Agent: DICE 1.8.20.1. > Content-Length: 0. > . > > # > U 94.94.X.X:1025 -> 178.79.x.x:5060 > ACK sip:526@test.domain:5060 SIP/2.0. > Via: SIP/2.0/UDP 94.94.X.X:5060;branch=z9hG4bK4a420cee;rport. > Max-Forwards: 70. > From: "asterisk" sip:512@test.domain;tag=as76007db0. > To: sip:526@test.domain:5060. > Contact: sip:512@94.94.X.X:5060. > Call-ID: 068a5a23639785a7583d952d6f9bca84@test.domain. > CSeq: 102 ACK. > User-Agent: DICE 1.8.20.1. > Content-Length: 0. > . > > # > T 178.79.x.x:5060 -> 94.94.X.X:61922 [AP] > ACK sip:526@94.94.X.X:1238;transport=TCP;ob SIP/2.0. > Via: SIP/2.0/TCP 178.79.x.x;branch=z9hG4bKcydzigwkX. > Via: SIP/2.0/UDP 94.94.X.X:5060;branch=z9hG4bK4a420cee;rport=1025. > Max-Forwards: 16. > From: "asterisk" sip:512@test.domain;tag=as76007db0. > To: sip:526@test.domain:5060. > Contact: sip:512@94.94.X.X:1025. > Call-ID: 068a5a23639785a7583d952d6f9bca84@test.domain. > CSeq: 102 ACK. > User-Agent: DICE 1.8.20.1. > Content-Length: 0. > . > > # > T 178.79.x.x:5060 -> 94.94.X.X:61922 [AP] > ACK sip:526@94.94.X.X:1238;transport=TCP;ob SIP/2.0. > Via: SIP/2.0/TCP 178.79.x.x;branch=z9hG4bKcydzigwkX. > Via: SIP/2.0/UDP 94.94.X.X:5060;branch=z9hG4bK4a420cee;rport=1025. > Max-Forwards: 16. > From: "asterisk" sip:512@test.domain;tag=as76007db0. > To: sip:526@test.domain:5060. > Contact: sip:512@94.94.X.X:1025. > Call-ID: 068a5a23639785a7583d952d6f9bca84@test.domain. > CSeq: 102 ACK. > User-Agent: DICE 1.8.20.1. > Content-Length: 0. > . > > >> Cheers, >> Daniel >>> Cheers, >>> Roberto Fichera. >>> >>>> Cheers, >>>> Daniel >>>> >>>> On 7/30/13 6:44 PM, Roberto Fichera wrote: >>>>> Hi All, >>>>> >>>>> Sorry for cross-posting this email to PJLIB, but maybe there are some things related. >>>>> Anyhow! I'm having problems on kamailio v4.0.2 under Fedora 18 64bit and TCP client like iPhone using PJSIP >>>>> as SIP >>>>> library. >>>>> Basically once the iPhone side in close the call (TCP->UDP) I'm getting the error below. Kamailio is running >>>>> under >>>>> a VPS >>>>> without >>>>> NATed network so it uses a real public address. Furthermore, note that tcp_main is answering to a >>>>> 192.168.2.98 ip >>>>> address >>>>> which is the iPhone client. This looks really strange to me since it should answer directly to the >>>>> public/port used >>>>> for >>>>> the registration >>>>> and not to a such kind of reserved address. The kamilio configuration is basically the default with a very few >>>>> changes >>>>> like NAT, rtpproxy and postgresql backend. >>>>> >>>>> This problems doesn't happen at all when using UDP->UDP calls. But I cannot use it because as you certain >>>>> know UDP >>>>> connection under iPhone will not work when the application run in background mode. >>>>> >>>>> Can someone suggest how to solve this issue or maybe suggest a TCP working solution for iPhone? >>>>> >>>>> Thanks in advance. >>>>> Roberto Fichera. >>>>> >>>>> Jul 30 16:21:53 proxy /usr/sbin/kamailio[9502]: ERROR: <core> [tcp_main.c:4432]: tcpconn_main_timeout(): connect >>>>> 192.168.2.98:5060 failed (timeout) >>>>> Jul 30 16:21:55 proxy rtpproxy[2262]: DBUG:get_command: received command "9483_9 D >>>>> 12d1d19926c4ff742a52f0c855b1bb83@94.94.x.x:5060 as74e0c388 GROahimCK6KTrl5CkYEg7nuoPIIXZ8cj" >>>>> Jul 30 16:21:55 proxy rtpproxy[2262]: INFO:handle_delete: forcefully deleting session 1 on ports 15604/17354 >>>>> Jul 30 16:21:55 proxy rtpproxy[2262]: INFO:remove_session: RTP stats: 354 in from callee, 603 in from caller, >>>>> 957 >>>>> relayed, 0 dropped >>>>> Jul 30 16:21:55 proxy rtpproxy[2262]: INFO:remove_session: RTCP stats: 5 in from callee, 2 in from caller, 7 >>>>> relayed, 0 >>>>> dropped >>>>> Jul 30 16:21:55 proxy rtpproxy[2262]: INFO:remove_session: session on ports 15604/17354 is cleaned up >>>>> Jul 30 16:21:55 proxy rtpproxy[2262]: DBUG:doreply: sending reply "9483_9 0 >>>>> Jul 30 16:21:55 proxy rtpproxy[2262]: " >>>>> Jul 30 16:22:04 proxy /usr/sbin/kamailio[9502]: ERROR: <core> [tcp_main.c:4432]: tcpconn_main_timeout(): connect >>>>> 192.168.2.98:5060 failed (timeout) >>>>> Jul 30 16:22:14 proxy rtpproxy[2262]: DBUG:get_command: received command "9496_16 D >>>>> 12d1d19926c4ff742a52f0c855b1bb83@94.94.x.x:5060 GROahimCK6KTrl5CkYEg7nuoPIIXZ8cj as74e0c388" >>>>> Jul 30 16:22:14 proxy rtpproxy[2262]: INFO:handle_command: delete request failed: session >>>>> 12d1d19926c4ff742a52f0c855b1bb83@94.94.x.x:5060, tags GROahimCK6KTrl5CkYEg7nuoPIIXZ8cj/as74e0c388 not found >>>>> Jul 30 16:22:14 proxy rtpproxy[2262]: DBUG:doreply: sending reply "9496_16 E8 >>>>> Jul 30 16:22:14 proxy rtpproxy[2262]: " >>>>> >>>>> >>>>> _______________________________________________ >>>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >>>>> sr-users@lists.sip-router.org >>>>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users@lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
On 08/14/2013 11:20 AM, Daniel-Constantin Mierla wrote:
Hi,
Hello,
On 8/14/13 11:16 AM, Roberto Fichera wrote:
On 08/14/2013 10:19 AM, Daniel-Constantin Mierla wrote:
Hi,
Hello,
as you can see in the REGISTER, the phone give a public IP where it can be contacted and kamailio tries to deliver to that address sip:528@94.94.X.X:1274;transport=TCP
REGISTER comes from another port, but that is allowed in SIP.
Yep! That's the problem
You should disable stun in the client and let the server care of nat traversal.
As already said all client doesn't use any stun-
then how they discover the public IP address if they are behind NAT? Or did I misunderstand that they are behind NAT?
Yes! They are behind NAT! Shouldn't kamailio detect such behaviour so it will change the contact header and the other fields accordingly using the nat_fix_XXXX functions?
If they are on the public IP, they they should listen on the port advertised in the Contact header of registration and accept the connection from Kamailio.
Cheers, Daniel
Or you can try to add flag 128 to nat test:
Also, you can try to add an alias for contact port:
Ok! I'll try both!
Cheers, Roberto Fichera.
Cheers, Daniel
On 8/13/13 4:17 PM, Roberto Fichera wrote:
On 08/13/2013 03:32 PM, Roberto Fichera wrote:
On 08/13/2013 03:25 PM, Daniel-Constantin Mierla wrote:
Can you get a ngrep trace for a registration as well (for the phone using tcp)?
Ok! I'll use pjsua from my local machine connecting in the same way as the TCP client was doing. The TCP client it's an iPhone using the same pjlib library.
I can confirm that the default cfg for TCP client doesn't work for me. My cfg file is attached. The TCP client doesn't receive any package at INVITE. Finally in /var/log/message I get this log below:
Aug 13 14:05:37 proxy /usr/sbin/kamailio[8401]: ERROR: <core> [tcp_main.c:4247]: handle_tcpconn_ev(): connect 94.94.X.X:1274 failed
Contact:: sip:528@94.94.X.X:1274;transport=TCP;ob;q=;expires=38;flags=0x0;cflags=0x0;state=0;socket=tcp:178.79.X.X:5060;methods=0x1FDF;user_agent=<PJSUA
v2.1-svn Linux-3.10.5.201/x86_64/glibc-2.17>;reg-id=0 [root@proxy ~]# kamctl ul show 512 Contact:: sip:512@94.94.X.X:5060;q=;expires=61;flags=0x0;cflags=0x40;state=0;socket=udp:178.79.X.X:5060;methods=0xFFFFFFFF;received=sip:94.94.X.X:1025
user_agent=<DICE 1.8.20.1>;reg-id=0
[root@proxy ~]# ngrep -W byline -d eth0 port 5060 interface: eth0 (178.79.X.X/255.255.255.0) filter: (ip or ip6) and ( port 5060 ) # T 94.94.X.X:49519 -> 178.79.x.x:5060 [AP] REGISTER sip:test.domain;transport=tcp;hide SIP/2.0. Via: SIP/2.0/TCP 94.94.X.X:49519;rport;branch=z9hG4bKPje03c11fd-d742-4502-98d8-69ca456ddd56;alias. Max-Forwards: 70. From: sip:528@test.domain;hide;tag=64c9e9b9-cb23-4a79-bb0d-2497135e449b. To: sip:528@test.domain;hide. Call-ID: 3ebadaeb-e6d6-42e3-9362-5ca4b5ddd8e1. CSeq: 30034 REGISTER. User-Agent: PJSUA v2.1-svn Linux-3.10.5.201/x86_64/glibc-2.17. Contact: sip:528@94.94.X.X:1274;transport=TCP;ob. Expires: 300. Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS. Content-Length: 0. .
# T 178.79.x.x:5060 -> 94.94.X.X:49519 [AP] SIP/2.0 401 Unauthorized. Via: SIP/2.0/TCP 94.94.X.X:49519;rport=49519;branch=z9hG4bKPje03c11fd-d742-4502-98d8-69ca456ddd56;alias. From: sip:528@test.domain;hide;tag=64c9e9b9-cb23-4a79-bb0d-2497135e449b. To: sip:528@test.domain;hide;tag=333a0370df4a40d5d5a0c21bb156e2a6.9167. Call-ID: 3ebadaeb-e6d6-42e3-9362-5ca4b5ddd8e1. CSeq: 30034 REGISTER. WWW-Authenticate: Digest realm="test.domain", nonce="Ugo+VFIKPSgVy8fh9fSy9SDLvT0wO4QV". Server: kamailio (4.0.2 (x86_64/linux)). Content-Length: 0. .
# U 178.79.x.x:5060 -> 94.94.X.X:1025 .... ## T 94.94.X.X:49519 -> 178.79.x.x:5060 [AP] REGISTER sip:test.domain;transport=tcp;hide SIP/2.0. Via: SIP/2.0/TCP 94.94.X.X:49519;rport;branch=z9hG4bKPj68a9c487-ae8c-428a-8c50-dd195a46a6bf;alias. Max-Forwards: 70. From: sip:528@test.domain;hide;tag=64c9e9b9-cb23-4a79-bb0d-2497135e449b. To: sip:528@test.domain;hide. Call-ID: 3ebadaeb-e6d6-42e3-9362-5ca4b5ddd8e1. CSeq: 30035 REGISTER. User-Agent: PJSUA v2.1-svn Linux-3.10.5.201/x86_64/glibc-2.17. Contact: sip:528@94.94.X.X:1274;transport=TCP;ob. Expires: 300. Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS. Authorization: Digest username="528", realm="test.domain", nonce="Ugo+VFIKPSgVy8fh9fSy9SDLvT0wO4QV", uri="sip:test.domain;transport=tcp;hide", response="ac1c7311ccb887fc8fb494d8ebf1bd36". Content-Length: 0. .
# T 178.79.x.x:5060 -> 94.94.X.X:49519 [AP] SIP/2.0 200 OK. Via: SIP/2.0/TCP 94.94.X.X:49519;rport=49519;branch=z9hG4bKPj68a9c487-ae8c-428a-8c50-dd195a46a6bf;alias. From: sip:528@test.domain;hide;tag=64c9e9b9-cb23-4a79-bb0d-2497135e449b. To: sip:528@test.domain;hide;tag=333a0370df4a40d5d5a0c21bb156e2a6.5440. Call-ID: 3ebadaeb-e6d6-42e3-9362-5ca4b5ddd8e1. CSeq: 30035 REGISTER. Contact: sip:528@94.94.X.X:1274;transport=TCP;ob;expires=300. Server: kamailio (4.0.2 (x86_64/linux)). Content-Length: 0. .
## U 94.94.X.X:1025 -> 178.79.x.x:5060 INVITE sip:528@test.domain:5060 SIP/2.0. Via: SIP/2.0/UDP 94.94.X.X:5060;branch=z9hG4bK69275d13;rport. Max-Forwards: 70. From: "asterisk" sip:512@test.domain;tag=as4531a4ee. To: sip:528@test.domain:5060. Contact: sip:512@94.94.X.X:5060. Call-ID: 2ea7e0f3173949d40b69385b62b79d6a@test.domain. CSeq: 102 INVITE. User-Agent: DICE 1.8.20.1. Date: Tue, 13 Aug 2013 14:04:11 GMT. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH. Supported: replaces, timer. Content-Type: application/sdp. Content-Length: 262. . v=0. o=root 883039875 883039875 IN IP4 94.94.X.X. s=Asterisk PBX 11.3.0. c=IN IP4 94.94.X.X. t=0 0. m=audio 18120 RTP/AVP 0 110 101. a=rtpmap:0 PCMU/8000. a=rtpmap:110 speex/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=sendrecv.
# U 178.79.x.x:5060 -> 94.94.X.X:1025 SIP/2.0 407 Proxy Authentication Required. Via: SIP/2.0/UDP 94.94.X.X:5060;branch=z9hG4bK69275d13;rport=1025. From: "asterisk" sip:512@test.domain;tag=as4531a4ee. To: sip:528@test.domain:5060;tag=333a0370df4a40d5d5a0c21bb156e2a6.038f. Call-ID: 2ea7e0f3173949d40b69385b62b79d6a@test.domain. CSeq: 102 INVITE. Proxy-Authenticate: Digest realm="test.domain", nonce="Ugo+XVIKPTFBd9tXC0QemfR9bHiZXO6x". Server: kamailio (4.0.2 (x86_64/linux)). Content-Length: 0. .
# U 94.94.X.X:1025 -> 178.79.x.x:5060 INVITE sip:528@test.domain:5060 SIP/2.0. Via: SIP/2.0/UDP 94.94.X.X:5060;branch=z9hG4bK69275d13;rport. Max-Forwards: 70. From: "asterisk" sip:512@test.domain;tag=as4531a4ee. To: sip:528@test.domain:5060. Contact: sip:512@94.94.X.X:5060. Call-ID: 2ea7e0f3173949d40b69385b62b79d6a@test.domain. CSeq: 102 INVITE. User-Agent: DICE 1.8.20.1. Date: Tue, 13 Aug 2013 14:04:11 GMT. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH. Supported: replaces, timer. Content-Type: application/sdp. Content-Length: 262. . v=0. o=root 883039875 883039875 IN IP4 94.94.X.X. s=Asterisk PBX 11.3.0. c=IN IP4 94.94.X.X. t=0 0. m=audio 18120 RTP/AVP 0 110 101. a=rtpmap:0 PCMU/8000. a=rtpmap:110 speex/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=sendrecv.
# U 178.79.x.x:5060 -> 94.94.X.X:1025 SIP/2.0 407 Proxy Authentication Required. Via: SIP/2.0/UDP 94.94.X.X:5060;branch=z9hG4bK69275d13;rport=1025. From: "asterisk" sip:512@test.domain;tag=as4531a4ee. To: sip:528@test.domain:5060;tag=333a0370df4a40d5d5a0c21bb156e2a6.038f. Call-ID: 2ea7e0f3173949d40b69385b62b79d6a@test.domain. CSeq: 102 INVITE. Proxy-Authenticate: Digest realm="test.domain", nonce="Ugo+XVIKPTFBd9tXC0QemfR9bHiZXO6x". Server: kamailio (4.0.2 (x86_64/linux)). Content-Length: 0. .
# U 94.94.X.X:1025 -> 178.79.x.x:5060 ACK sip:528@test.domain:5060 SIP/2.0. Via: SIP/2.0/UDP 94.94.X.X:5060;branch=z9hG4bK69275d13;rport. Max-Forwards: 70. From: "asterisk" sip:512@test.domain;tag=as4531a4ee. To: sip:528@test.domain:5060;tag=333a0370df4a40d5d5a0c21bb156e2a6.038f. Contact: sip:512@94.94.X.X:5060. Call-ID: 2ea7e0f3173949d40b69385b62b79d6a@test.domain. CSeq: 102 ACK. User-Agent: DICE 1.8.20.1. Content-Length: 0. .
# U 94.94.X.X:1025 -> 178.79.x.x:5060 ACK sip:528@test.domain:5060 SIP/2.0. Via: SIP/2.0/UDP 94.94.X.X:5060;branch=z9hG4bK69275d13;rport. Max-Forwards: 70. From: "asterisk" sip:512@test.domain;tag=as4531a4ee. To: sip:528@test.domain:5060. Contact: sip:512@94.94.X.X:5060. Call-ID: 2ea7e0f3173949d40b69385b62b79d6a@test.domain. CSeq: 102 ACK. User-Agent: DICE 1.8.20.1. Content-Length: 0. .
------------------ This is the client behind NAT ---------------------------
REGISTER sip:test.domain;transport=tcp;hide SIP/2.0 Via: SIP/2.0/TCP 192.168.2.90:49519;rport;branch=z9hG4bKPj92d04d45-bd84-45b3-9439-563ebfaebf00;alias Max-Forwards: 70 From: sip:528@test.domain;hide;tag=ab763476-0fdb-4d85-9064-1deabcaa23e4 To: sip:528@test.domain;hide Call-ID: 3ebadaeb-e6d6-42e3-9362-5ca4b5ddd8e1 CSeq: 30028 REGISTER User-Agent: PJSUA v2.1-svn Linux-3.10.5.201/x86_64/glibc-2.17 Contact: sip:528@192.168.2.90:5060;transport=TCP;ob Expires: 300 Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Content-Length: 0
--end msg-- 15:54:02.580 pjsua_acc.c .Acc 0: Registration sent
> 15:54:02.681 tcpc0x1515a08 !TCP transport 192.168.2.90:49519 is connected to 178.79.x.x:5060
15:54:02.681 pjsua_app.c SIP TCP transport is connected to [178.79.x.x:5060] 15:54:02.785 pjsua_core.c .RX 538 bytes Response msg 401/REGISTER/cseq=30028 (rdata0x1515cf8) from TCP 178.79.x.x:5060: SIP/2.0 401 Unauthorized Via: SIP/2.0/TCP 192.168.2.90:49519;rport=49519;branch=z9hG4bKPj92d04d45-bd84-45b3-9439-563ebfaebf00;alias From: sip:528@test.domain;hide;tag=ab763476-0fdb-4d85-9064-1deabcaa23e4 To: sip:528@test.domain;hide;tag=333a0370df4a40d5d5a0c21bb156e2a6.9663 Call-ID: 3ebadaeb-e6d6-42e3-9362-5ca4b5ddd8e1 CSeq: 30028 REGISTER WWW-Authenticate: Digest realm="test.domain", nonce="Ugo7/FIKOtBneWAQzD/VoJuucFT3cNuF" Server: kamailio (4.0.2 (x86_64/linux)) Content-Length: 0
--end msg-- 15:54:02.785 pjsua_core.c ....TX 849 bytes Request msg REGISTER/cseq=30029 (tdta0x1513000) to TCP 178.79.x.x:5060: REGISTER sip:test.domain;transport=tcp;hide SIP/2.0 Via: SIP/2.0/TCP 192.168.2.90:49519;rport;branch=z9hG4bKPj03a4a7eb-d086-4add-b84c-ca5c6cf4d05d;alias Max-Forwards: 70 From: sip:528@test.domain;hide;tag=ab763476-0fdb-4d85-9064-1deabcaa23e4 To: sip:528@test.domain;hide Call-ID: 3ebadaeb-e6d6-42e3-9362-5ca4b5ddd8e1 CSeq: 30029 REGISTER User-Agent: PJSUA v2.1-svn Linux-3.10.5.201/x86_64/glibc-2.17 Contact: sip:528@192.168.2.90:5060;transport=TCP;ob Expires: 300 Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Authorization: Digest username="528", realm="test.domain", nonce="Ugo7/FIKOtBneWAQzD/VoJuucFT3cNuF", uri="sip:test.domain;transport=tcp;hide", response="c3dd687602ec35b2403b3eb142b496f5" Content-Length: 0
--end msg-- 15:54:02.890 pjsua_core.c .RX 495 bytes Response msg 200/REGISTER/cseq=30029 (rdata0x1515cf8) from TCP 178.79.x.x:5060: SIP/2.0 200 OK Via: SIP/2.0/TCP 192.168.2.90:49519;rport=49519;branch=z9hG4bKPj03a4a7eb-d086-4add-b84c-ca5c6cf4d05d;alias From: sip:528@test.domain;hide;tag=ab763476-0fdb-4d85-9064-1deabcaa23e4 To: sip:528@test.domain;hide;tag=333a0370df4a40d5d5a0c21bb156e2a6.189b Call-ID: 3ebadaeb-e6d6-42e3-9362-5ca4b5ddd8e1 CSeq: 30029 REGISTER Contact: sip:528@192.168.2.90:5060;transport=TCP;ob;expires=300 Server: kamailio (4.0.2 (x86_64/linux)) Content-Length: 0
--end msg-- 15:54:02.890 pjsua_acc.c ....SIP outbound status for acc 0 is not active 15:54:02.890 pjsua_acc.c ....sip:528@test.domain;transport=tcp;hide: registration success, status=200 (OK), will re-register in 300 seconds
Daniel
On 8/13/13 3:23 PM, Roberto Fichera wrote: > On 08/13/2013 03:15 PM, Roberto Fichera wrote: >> On 08/13/2013 02:33 PM, Daniel-Constantin Mierla wrote: >>> Hello, >>> >>> On 8/13/13 1:10 PM, Roberto Fichera wrote: >>>> On 08/13/2013 12:03 PM, Daniel-Constantin Mierla wrote: >>>>> Hello, >>>>> >>>>> you should grab the ngrep for such call to understand better what happens. Also, dumping the location records >>>>> will be >>>>> useful (kamctl ul show). >>>>> >>>>> Also, be sure that tcp connection lifetime is long enough to survive re-registration. To avoid trying to open >>>>> connections behind nat, use set_forward_no_connect() for calls involving nat traversal. >>>> I'm using the default conf coming from fedora rpm. So, mainly the problem seems related to kamailio >>>> which doesn't reuse the TCP port used by NATed clients. I've also notice that the received >>>> field isn't set at all, so this means that the contact will not get aliased at all. >>>> >>>> I would really like to have a look to a working cfg file for TCP NATed clients that reuse the TCP port. >>>> Even better if the configuration is based on the fedora default rpm. >>> if received is not set, then means the register was not detected as coming from behind nat. Is the phone using >>> stun? >> I'm testing with a normal rtpproxy configuration. BTW udp -> udp work perfectly. >> >>> Again, put here the ngrep for registration and a call to see if something is wrong with signaling. There is no >>> help >>> that we can provide otherwise. The default config works fine for tcp and natted clients, I use it everywhere for >>> this >>> case without issues. >> I tried the default cfg enabling both NAT and RTPproxy, but seems that kamailio doesn't reuse TCP ports. >> Anyway, this is a call from UDP (512) -> TCP (526) both behind the same NAT, from kamailio point of view > I forgot to say that the received field is now present because I've changed the > route[NATDETECT] in the default configuration as > > route[NATDETECT] { > #!ifdef WITH_NAT > force_rport(); > > -->>> if (nat_uac_test("19") || proto != UDP) { > if (is_method("REGISTER")) { > fix_nated_register(); > } else { > fix_nated_contact(); > } > setflag(FLT_NATS); > } > #!endif > return; > } > > >> [root@proxy ~]# kamctl ul show 526 >> Contact:: >> sip:526@94.94.X.X:1238;transport=TCP;ob;q=;expires=537;flags=0x0;cflags=0x40;state=0;socket=tcp:178.79.x.x:5060;methods=0x1FDF;received=sip:94.94.X.X:61922;transport=TCP;user_agent=<DICE >> >> >> >> Smartphone 1.0/iPhone>;reg-id=0 >> [root@proxy ~]# kamctl ul show 512 >> Contact:: >> sip:512@94.94.X.X:5060;q=;expires=32;flags=0x0;cflags=0x40;state=0;socket=udp:178.79.x.x:5060;methods=0xFFFFFFFF;received=sip:94.94.X.X:1025;user_agent=<DICE >> >> >> >> 1.8.20.1>;reg-id=0 >> [root@proxy ~]# >> >> >> U 94.94.X.X:1025 -> 178.79.x.x:5060 >> INVITE sip:526@test.domain:5060 SIP/2.0. >> Via: SIP/2.0/UDP 94.94.X.X:5060;branch=z9hG4bK4a420cee;rport. >> Max-Forwards: 70. >> From: "asterisk" sip:512@test.domain;tag=as76007db0. >> To: sip:526@test.domain:5060. >> Contact: sip:512@94.94.X.X:5060. >> Call-ID: 068a5a23639785a7583d952d6f9bca84@test.domain. >> CSeq: 102 INVITE. >> User-Agent: DICE 1.8.20.1. >> Date: Tue, 13 Aug 2013 13:04:30 GMT. >> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH. >> Supported: replaces, timer. >> Content-Type: application/sdp. >> Content-Length: 264. >> . >> v=0. >> o=root 1263161426 1263161426 IN IP4 94.94.X.X. >> s=Asterisk PBX 11.3.0. >> c=IN IP4 94.94.X.X. >> t=0 0. >> m=audio 10782 RTP/AVP 0 110 101. >> a=rtpmap:0 PCMU/8000. >> a=rtpmap:110 speex/8000. >> a=rtpmap:101 telephone-event/8000. >> a=fmtp:101 0-16. >> a=ptime:20. >> a=sendrecv. >> >> # >> U 178.79.x.x:5060 -> 94.94.X.X:1025 >> SIP/2.0 407 Proxy Authentication Required. >> Via: SIP/2.0/UDP 94.94.X.X:5060;branch=z9hG4bK4a420cee;rport=1025. >> From: "asterisk" sip:512@test.domain;tag=as76007db0. >> To: sip:526@test.domain:5060;tag=333a0370df4a40d5d5a0c21bb156e2a6.4a00. >> Call-ID: 068a5a23639785a7583d952d6f9bca84@test.domain. >> CSeq: 102 INVITE. >> Proxy-Authenticate: Digest realm="test.domain", nonce="UgowYFIKLzQXDSUWUpy4xlhdXZ7WNjPt". >> Server: kamailio (4.0.2 (x86_64/linux)). >> Content-Length: 0. >> . >> >> # >> U 94.94.X.X:1025 -> 178.79.x.x:5060 >> INVITE sip:526@test.domain:5060 SIP/2.0. >> Via: SIP/2.0/UDP 94.94.X.X:5060;branch=z9hG4bK4a420cee;rport. >> Max-Forwards: 70. >> From: "asterisk" sip:512@test.domain;tag=as76007db0. >> To: sip:526@test.domain:5060. >> Contact: sip:512@94.94.X.X:5060. >> Call-ID: 068a5a23639785a7583d952d6f9bca84@test.domain. >> CSeq: 102 INVITE. >> User-Agent: DICE 1.8.20.1. >> Date: Tue, 13 Aug 2013 13:04:30 GMT. >> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH. >> Supported: replaces, timer. >> Content-Type: application/sdp. >> Content-Length: 264. >> . >> v=0. >> o=root 1263161426 1263161426 IN IP4 94.94.X.X. >> s=Asterisk PBX 11.3.0. >> c=IN IP4 94.94.X.X. >> t=0 0. >> m=audio 10782 RTP/AVP 0 110 101. >> a=rtpmap:0 PCMU/8000. >> a=rtpmap:110 speex/8000. >> a=rtpmap:101 telephone-event/8000. >> a=fmtp:101 0-16. >> a=ptime:20. >> a=sendrecv. >> >> # >> U 178.79.x.x:5060 -> 94.94.X.X:1025 >> SIP/2.0 407 Proxy Authentication Required. >> Via: SIP/2.0/UDP 94.94.X.X:5060;branch=z9hG4bK4a420cee;rport=1025. >> From: "asterisk" sip:512@test.domain;tag=as76007db0. >> To: sip:526@test.domain:5060;tag=333a0370df4a40d5d5a0c21bb156e2a6.4a00. >> Call-ID: 068a5a23639785a7583d952d6f9bca84@test.domain. >> CSeq: 102 INVITE. >> Proxy-Authenticate: Digest realm="test.domain", nonce="UgowYFIKLzQXDSUWUpy4xlhdXZ7WNjPt". >> Server: kamailio (4.0.2 (x86_64/linux)). >> Content-Length: 0. >> . >> >> # >> U 94.94.X.X:1025 -> 178.79.x.x:5060 >> ACK sip:526@test.domain:5060 SIP/2.0. >> Via: SIP/2.0/UDP 94.94.X.X:5060;branch=z9hG4bK4a420cee;rport. >> Max-Forwards: 70. >> From: "asterisk" sip:512@test.domain;tag=as76007db0. >> To: sip:526@test.domain:5060;tag=333a0370df4a40d5d5a0c21bb156e2a6.4a00. >> Contact: sip:512@94.94.X.X:5060. >> Call-ID: 068a5a23639785a7583d952d6f9bca84@test.domain. >> CSeq: 102 ACK. >> User-Agent: DICE 1.8.20.1. >> Content-Length: 0. >> . >> >> # >> U 94.94.X.X:1025 -> 178.79.x.x:5060 >> ACK sip:526@test.domain:5060 SIP/2.0. >> Via: SIP/2.0/UDP 94.94.X.X:5060;branch=z9hG4bK4a420cee;rport. >> Max-Forwards: 70. >> From: "asterisk" sip:512@test.domain;tag=as76007db0. >> To: sip:526@test.domain:5060. >> Contact: sip:512@94.94.X.X:5060. >> Call-ID: 068a5a23639785a7583d952d6f9bca84@test.domain. >> CSeq: 102 ACK. >> User-Agent: DICE 1.8.20.1. >> Content-Length: 0. >> . >> >> # >> T 178.79.x.x:5060 -> 94.94.X.X:61922 [AP] >> ACK sip:526@94.94.X.X:1238;transport=TCP;ob SIP/2.0. >> Via: SIP/2.0/TCP 178.79.x.x;branch=z9hG4bKcydzigwkX. >> Via: SIP/2.0/UDP 94.94.X.X:5060;branch=z9hG4bK4a420cee;rport=1025. >> Max-Forwards: 16. >> From: "asterisk" sip:512@test.domain;tag=as76007db0. >> To: sip:526@test.domain:5060. >> Contact: sip:512@94.94.X.X:1025. >> Call-ID: 068a5a23639785a7583d952d6f9bca84@test.domain. >> CSeq: 102 ACK. >> User-Agent: DICE 1.8.20.1. >> Content-Length: 0. >> . >> >> # >> T 178.79.x.x:5060 -> 94.94.X.X:61922 [AP] >> ACK sip:526@94.94.X.X:1238;transport=TCP;ob SIP/2.0. >> Via: SIP/2.0/TCP 178.79.x.x;branch=z9hG4bKcydzigwkX. >> Via: SIP/2.0/UDP 94.94.X.X:5060;branch=z9hG4bK4a420cee;rport=1025. >> Max-Forwards: 16. >> From: "asterisk" sip:512@test.domain;tag=as76007db0. >> To: sip:526@test.domain:5060. >> Contact: sip:512@94.94.X.X:1025. >> Call-ID: 068a5a23639785a7583d952d6f9bca84@test.domain. >> CSeq: 102 ACK. >> User-Agent: DICE 1.8.20.1. >> Content-Length: 0. >> . >> >> >>> Cheers, >>> Daniel >>>> Cheers, >>>> Roberto Fichera. >>>> >>>>> Cheers, >>>>> Daniel >>>>> >>>>> On 7/30/13 6:44 PM, Roberto Fichera wrote: >>>>>> Hi All, >>>>>> >>>>>> Sorry for cross-posting this email to PJLIB, but maybe there are some things related. >>>>>> Anyhow! I'm having problems on kamailio v4.0.2 under Fedora 18 64bit and TCP client like iPhone using PJSIP >>>>>> as SIP >>>>>> library. >>>>>> Basically once the iPhone side in close the call (TCP->UDP) I'm getting the error below. Kamailio is running >>>>>> under >>>>>> a VPS >>>>>> without >>>>>> NATed network so it uses a real public address. Furthermore, note that tcp_main is answering to a >>>>>> 192.168.2.98 ip >>>>>> address >>>>>> which is the iPhone client. This looks really strange to me since it should answer directly to the >>>>>> public/port used >>>>>> for >>>>>> the registration >>>>>> and not to a such kind of reserved address. The kamilio configuration is basically the default with a very few >>>>>> changes >>>>>> like NAT, rtpproxy and postgresql backend. >>>>>> >>>>>> This problems doesn't happen at all when using UDP->UDP calls. But I cannot use it because as you certain >>>>>> know UDP >>>>>> connection under iPhone will not work when the application run in background mode. >>>>>> >>>>>> Can someone suggest how to solve this issue or maybe suggest a TCP working solution for iPhone? >>>>>> >>>>>> Thanks in advance. >>>>>> Roberto Fichera. >>>>>> >>>>>> Jul 30 16:21:53 proxy /usr/sbin/kamailio[9502]: ERROR: <core> [tcp_main.c:4432]: tcpconn_main_timeout(): >>>>>> connect >>>>>> 192.168.2.98:5060 failed (timeout) >>>>>> Jul 30 16:21:55 proxy rtpproxy[2262]: DBUG:get_command: received command "9483_9 D >>>>>> 12d1d19926c4ff742a52f0c855b1bb83@94.94.x.x:5060 as74e0c388 GROahimCK6KTrl5CkYEg7nuoPIIXZ8cj" >>>>>> Jul 30 16:21:55 proxy rtpproxy[2262]: INFO:handle_delete: forcefully deleting session 1 on ports 15604/17354 >>>>>> Jul 30 16:21:55 proxy rtpproxy[2262]: INFO:remove_session: RTP stats: 354 in from callee, 603 in from caller, >>>>>> 957 >>>>>> relayed, 0 dropped >>>>>> Jul 30 16:21:55 proxy rtpproxy[2262]: INFO:remove_session: RTCP stats: 5 in from callee, 2 in from caller, 7 >>>>>> relayed, 0 >>>>>> dropped >>>>>> Jul 30 16:21:55 proxy rtpproxy[2262]: INFO:remove_session: session on ports 15604/17354 is cleaned up >>>>>> Jul 30 16:21:55 proxy rtpproxy[2262]: DBUG:doreply: sending reply "9483_9 0 >>>>>> Jul 30 16:21:55 proxy rtpproxy[2262]: " >>>>>> Jul 30 16:22:04 proxy /usr/sbin/kamailio[9502]: ERROR: <core> [tcp_main.c:4432]: tcpconn_main_timeout(): >>>>>> connect >>>>>> 192.168.2.98:5060 failed (timeout) >>>>>> Jul 30 16:22:14 proxy rtpproxy[2262]: DBUG:get_command: received command "9496_16 D >>>>>> 12d1d19926c4ff742a52f0c855b1bb83@94.94.x.x:5060 GROahimCK6KTrl5CkYEg7nuoPIIXZ8cj as74e0c388" >>>>>> Jul 30 16:22:14 proxy rtpproxy[2262]: INFO:handle_command: delete request failed: session >>>>>> 12d1d19926c4ff742a52f0c855b1bb83@94.94.x.x:5060, tags GROahimCK6KTrl5CkYEg7nuoPIIXZ8cj/as74e0c388 not found >>>>>> Jul 30 16:22:14 proxy rtpproxy[2262]: DBUG:doreply: sending reply "9496_16 E8 >>>>>> Jul 30 16:22:14 proxy rtpproxy[2262]: " >>>>>> >>>>>> >>>>>> _______________________________________________ >>>>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >>>>>> sr-users@lists.sip-router.org >>>>>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >> _______________________________________________ >> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >> sr-users@lists.sip-router.org >> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >>
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Hello,
On 8/14/13 11:26 AM, Roberto Fichera wrote:
On 08/14/2013 11:20 AM, Daniel-Constantin Mierla wrote:
Hi,
Hello,
On 8/14/13 11:16 AM, Roberto Fichera wrote:
On 08/14/2013 10:19 AM, Daniel-Constantin Mierla wrote:
Hi,
Hello,
as you can see in the REGISTER, the phone give a public IP where it can be contacted and kamailio tries to deliver to that address sip:528@94.94.X.X:1274;transport=TCP
REGISTER comes from another port, but that is allowed in SIP.
Yep! That's the problem
You should disable stun in the client and let the server care of nat traversal.
As already said all client doesn't use any stun-
then how they discover the public IP address if they are behind NAT? Or did I misunderstand that they are behind NAT?
Yes! They are behind NAT! Shouldn't kamailio detect such behaviour so it will change the contact header and the other fields accordingly using the nat_fix_XXXX functions?
detection of whether a device is behind nat is done based on comparing source IP of the packet with the IP addresses in the SIP message. It is no other way from application point of view.
If you look at the trace, the REGISTER has only public IP addresses, no private address. Also, the Via header is perfectly matching the the source IP and port of the packet.
Therefore, if the device is behind a nat in a private network, how does it discover the public IP or the NAT router? Is the nat router running a SIP ALG? Can you get the SIP trace on the device itself? Cheers, Daniel
On 08/14/2013 11:31 AM, Daniel-Constantin Mierla wrote:
Hi,
Hello,
On 8/14/13 11:26 AM, Roberto Fichera wrote:
On 08/14/2013 11:20 AM, Daniel-Constantin Mierla wrote:
Hi,
Hello,
On 8/14/13 11:16 AM, Roberto Fichera wrote:
On 08/14/2013 10:19 AM, Daniel-Constantin Mierla wrote:
Hi,
Hello,
as you can see in the REGISTER, the phone give a public IP where it can be contacted and kamailio tries to deliver to that address sip:528@94.94.X.X:1274;transport=TCP
REGISTER comes from another port, but that is allowed in SIP.
Yep! That's the problem
You should disable stun in the client and let the server care of nat traversal.
As already said all client doesn't use any stun-
then how they discover the public IP address if they are behind NAT? Or did I misunderstand that they are behind NAT?
Yes! They are behind NAT! Shouldn't kamailio detect such behaviour so it will change the contact header and the other fields accordingly using the nat_fix_XXXX functions?
detection of whether a device is behind nat is done based on comparing source IP of the packet with the IP addresses in the SIP message. It is no other way from application point of view.
If you look at the trace, the REGISTER has only public IP addresses, no private address. Also, the Via header is perfectly matching the the source IP and port of the packet.
Mhhh!!! That's true!
Therefore, if the device is behind a nat in a private network, how does it discover the public IP or the NAT router? Is the nat router running a SIP ALG? Can you get the SIP trace on the device itself?
Good question! It's a cisco 877 but the conf doesn't have any kind of SIP related configuration. Does it handle transparently the SIP traffic?
Cisco IOS Software, C870 Software (C870-ADVIPSERVICESK9-M), Version 12.4(15)T5, RELEASE SOFTWARE (fc4) Technical Support: http://www.cisco.com/techsupport Copyright (c) 1986-2008 by Cisco Systems, Inc. Compiled Thu 01-May-08 02:31 by prod_rel_team
Cheers, Daniel
On 08/14/2013 11:51 AM, Roberto Fichera wrote:
On 08/14/2013 11:31 AM, Daniel-Constantin Mierla wrote:
Hi,
Hello,
On 8/14/13 11:26 AM, Roberto Fichera wrote:
On 08/14/2013 11:20 AM, Daniel-Constantin Mierla wrote:
Hi,
Hello,
On 8/14/13 11:16 AM, Roberto Fichera wrote:
On 08/14/2013 10:19 AM, Daniel-Constantin Mierla wrote:
Hi,
Hello,
as you can see in the REGISTER, the phone give a public IP where it can be contacted and kamailio tries to deliver to that address sip:528@94.94.X.X:1274;transport=TCP
REGISTER comes from another port, but that is allowed in SIP.
Yep! That's the problem
You should disable stun in the client and let the server care of nat traversal.
As already said all client doesn't use any stun-
then how they discover the public IP address if they are behind NAT? Or did I misunderstand that they are behind NAT?
Yes! They are behind NAT! Shouldn't kamailio detect such behaviour so it will change the contact header and the other fields accordingly using the nat_fix_XXXX functions?
detection of whether a device is behind nat is done based on comparing source IP of the packet with the IP addresses in the SIP message. It is no other way from application point of view.
If you look at the trace, the REGISTER has only public IP addresses, no private address. Also, the Via header is perfectly matching the the source IP and port of the packet.
Mhhh!!! That's true!
Therefore, if the device is behind a nat in a private network, how does it discover the public IP or the NAT router? Is the nat router running a SIP ALG? Can you get the SIP trace on the device itself?
Good question! It's a cisco 877 but the conf doesn't have any kind of SIP related configuration. Does it handle transparently the SIP traffic?
Cisco IOS Software, C870 Software (C870-ADVIPSERVICESK9-M), Version 12.4(15)T5, RELEASE SOFTWARE (fc4) Technical Support: http://www.cisco.com/techsupport Copyright (c) 1986-2008 by Cisco Systems, Inc. Compiled Thu 01-May-08 02:31 by prod_rel_team
Ok! There are two commands to disable it, since it's enabled by default!!!
no ip nat service sip tcp port 5060 no ip nat service sip udp port 5060
I'll test it right now!
Cheers, Daniel
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
If you won't be able to disable SIP ALG on your router you can fill $avp(received) manually before calling save(): $avp(received) = "sip:" + $si + ":" + $sp + ";transport=" + $proto;
In this case all user location records will have the "received" attribut even if a UA isn't behind NAT, but I don't see any problems with that.
On 08/14/2013 11:51 AM, Roberto Fichera wrote:
On 08/14/2013 11:31 AM, Daniel-Constantin Mierla wrote:
Hi,
Hello,
On 8/14/13 11:26 AM, Roberto Fichera wrote:
On 08/14/2013 11:20 AM, Daniel-Constantin Mierla wrote:
Hi,
Hello,
On 8/14/13 11:16 AM, Roberto Fichera wrote:
On 08/14/2013 10:19 AM, Daniel-Constantin Mierla wrote:
Hi,
> Hello, > > as you can see in the REGISTER, the phone give a public IP where it can be contacted and kamailio tries to deliver to > that address sip:528@94.94.X.X:1274;transport=TCP > > REGISTER comes from another port, but that is allowed in SIP. Yep! That's the problem
> You should disable stun in the client and let the server care of nat traversal. As already said all client doesn't use any stun-
then how they discover the public IP address if they are behind NAT? Or did I misunderstand that they are behind NAT?
Yes! They are behind NAT! Shouldn't kamailio detect such behaviour so it will change the contact header and the other fields accordingly using the nat_fix_XXXX functions?
detection of whether a device is behind nat is done based on comparing source IP of the packet with the IP addresses in the SIP message. It is no other way from application point of view.
If you look at the trace, the REGISTER has only public IP addresses, no private address. Also, the Via header is perfectly matching the the source IP and port of the packet.
Mhhh!!! That's true!
Therefore, if the device is behind a nat in a private network, how does it discover the public IP or the NAT router? Is the nat router running a SIP ALG? Can you get the SIP trace on the device itself?
Good question! It's a cisco 877 but the conf doesn't have any kind of SIP related configuration. Does it handle transparently the SIP traffic?
Cisco IOS Software, C870 Software (C870-ADVIPSERVICESK9-M), Version 12.4(15)T5, RELEASE SOFTWARE (fc4) Technical Support: http://www.cisco.com/techsupport Copyright (c) 1986-2008 by Cisco Systems, Inc. Compiled Thu 01-May-08 02:31 by prod_rel_team
Ok! There are two commands to disable it, since it's enabled by default!!!
no ip nat service sip tcp port 5060 no ip nat service sip udp port 5060
I'll test it right now!
Cheers, Daniel
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
On 08/14/2013 04:36 PM, Vitaliy Aleksandrov wrote:
If you won't be able to disable SIP ALG on your router you can fill $avp(received) manually before calling save(): $avp(received) = "sip:" + $si + ":" + $sp + ";transport=" + $proto;
In this case all user location records will have the "received" attribut even if a UA isn't behind NAT, but I don't see any problems with that.
This one looks working, but the callee doesn't answer correctly because the TCP isn't correct:
Contact:: sip:528@94.94.X.X:1380;transport=TCP;ob;q=;expires=294;flags=0x0;cflags=0x0;state=0;socket=tcp:178.79.X.X:5060;methods=0x1FDF;received=sip:94.94.X.X:37030;transport=tcp;user_agent=<PJSUA v2.1-svn Linux-3.10.5.201/x86_64/glibc-2.17>;reg-id=0
The contact uses a port which isn't translated inside by the router, the received field shows the right one. Should I change the Contact header instead? In case, how can I do that?
Cheers, Roberto Fichera.
On 08/14/2013 11:51 AM, Roberto Fichera wrote:
On 08/14/2013 11:31 AM, Daniel-Constantin Mierla wrote:
Hi,
Hello,
On 8/14/13 11:26 AM, Roberto Fichera wrote:
On 08/14/2013 11:20 AM, Daniel-Constantin Mierla wrote:
Hi,
Hello,
On 8/14/13 11:16 AM, Roberto Fichera wrote: > On 08/14/2013 10:19 AM, Daniel-Constantin Mierla wrote: > > Hi, > >> Hello, >> >> as you can see in the REGISTER, the phone give a public IP where it can be contacted and kamailio tries to >> deliver to >> that address sip:528@94.94.X.X:1274;transport=TCP >> >> REGISTER comes from another port, but that is allowed in SIP. > Yep! That's the problem > >> You should disable stun in the client and let the server care of nat traversal. > As already said all client doesn't use any stun- then how they discover the public IP address if they are behind NAT? Or did I misunderstand that they are behind NAT?
Yes! They are behind NAT! Shouldn't kamailio detect such behaviour so it will change the contact header and the other fields accordingly using the nat_fix_XXXX functions?
detection of whether a device is behind nat is done based on comparing source IP of the packet with the IP addresses in the SIP message. It is no other way from application point of view.
If you look at the trace, the REGISTER has only public IP addresses, no private address. Also, the Via header is perfectly matching the the source IP and port of the packet.
Mhhh!!! That's true!
Therefore, if the device is behind a nat in a private network, how does it discover the public IP or the NAT router? Is the nat router running a SIP ALG? Can you get the SIP trace on the device itself?
Good question! It's a cisco 877 but the conf doesn't have any kind of SIP related configuration. Does it handle transparently the SIP traffic?
Cisco IOS Software, C870 Software (C870-ADVIPSERVICESK9-M), Version 12.4(15)T5, RELEASE SOFTWARE (fc4) Technical Support: http://www.cisco.com/techsupport Copyright (c) 1986-2008 by Cisco Systems, Inc. Compiled Thu 01-May-08 02:31 by prod_rel_team
Ok! There are two commands to disable it, since it's enabled by default!!!
no ip nat service sip tcp port 5060 no ip nat service sip udp port 5060
I'll test it right now!
Cheers, Daniel
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
On 08/14/2013 07:32 PM, Roberto Fichera wrote:
On 08/14/2013 04:36 PM, Vitaliy Aleksandrov wrote:
If you won't be able to disable SIP ALG on your router you can fill $avp(received) manually before calling save(): $avp(received) = "sip:" + $si + ":" + $sp + ";transport=" + $proto;
In this case all user location records will have the "received" attribut even if a UA isn't behind NAT, but I don't see any problems with that.
This one looks working, but the callee doesn't answer correctly because the TCP isn't correct:
Contact:: sip:528@94.94.X.X:1380;transport=TCP;ob;q=;expires=294;flags=0x0;cflags=0x0;state=0;socket=tcp:178.79.X.X:5060;methods=0x1FDF;received=sip:94.94.X.X:37030;transport=tcp;user_agent=<PJSUA v2.1-svn Linux-3.10.5.201/x86_64/glibc-2.17>;reg-id=0
The contact uses a port which isn't translated inside by the router, the received field shows the right one. Should I change the Contact header instead? In case, how can I do that?
I didn't understand what "callee doesn't answer correctly" means. Callee doesn't know what's in the received field of its registration. The only problem I see is that your router changes ip in the contact field of REGISTER requests and then kamailio puts this value (new_ip:old_port) to INVITEs destined to UAs behind your NAT router.
It's not likely, but maybe pjsip doesn't like INVITEs with RURI which differs from what it put to the contact during registration. IIRC pjsua prints lots of debugging information. So you can check if pjsua shows anything when INVITE comes in its log and also attach a trace of such a call.
P.S. IMHO the best way to pass sip through such routers is sip over TLS.
On 08/14/2013 09:58 PM, Vitaliy Aleksandrov wrote:
On 08/14/2013 07:32 PM, Roberto Fichera wrote:
On 08/14/2013 04:36 PM, Vitaliy Aleksandrov wrote:
If you won't be able to disable SIP ALG on your router you can fill $avp(received) manually before calling save(): $avp(received) = "sip:" + $si + ":" + $sp + ";transport=" + $proto;
In this case all user location records will have the "received" attribut even if a UA isn't behind NAT, but I don't see any problems with that.
This one looks working, but the callee doesn't answer correctly because the TCP isn't correct:
Contact:: sip:528@94.94.X.X:1380;transport=TCP;ob;q=;expires=294;flags=0x0;cflags=0x0;state=0;socket=tcp:178.79.X.X:5060;methods=0x1FDF;received=sip:94.94.X.X:37030;transport=tcp;user_agent=<PJSUA
v2.1-svn Linux-3.10.5.201/x86_64/glibc-2.17>;reg-id=0
The contact uses a port which isn't translated inside by the router, the received field shows the right one. Should I change the Contact header instead? In case, how can I do that?
I didn't understand what "callee doesn't answer correctly" means. Callee doesn't know what's in the received field of its registration.
Sorry! My typo! I meant why kamilio wasn't reusing the TCP port specified in the REGISTER even for the INVITE. I mean, kamailio now knows the received field so I'm expecting it routes the requests for the called UA through the address:port specified in this field.
The only problem I see is that your router changes ip in the contact field of REGISTER requests and then kamailio puts this value (new_ip:old_port) to INVITEs destined to UAs behind your NAT router.
Indeed, my testing router is affected by this ALG "problem"! I guess it's better to disable it and complete simple TCP tests then move to TLS.
It's not likely, but maybe pjsip doesn't like INVITEs with RURI which differs from what it put to the contact during registration. IIRC pjsua prints lots of debugging information. So you can check if pjsua shows anything when INVITE comes in its log and also attach a trace of such a call.
pjsua doesn't report any problem so far, so seems accepting the INVITE.
P.S. IMHO the best way to pass sip through such routers is sip over TLS.
Yep! I'll do that.
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
On 08/16/2013 12:29 PM, Roberto Fichera wrote:
On 08/14/2013 09:58 PM, Vitaliy Aleksandrov wrote:
On 08/14/2013 07:32 PM, Roberto Fichera wrote:
On 08/14/2013 04:36 PM, Vitaliy Aleksandrov wrote:
If you won't be able to disable SIP ALG on your router you can fill $avp(received) manually before calling save(): $avp(received) = "sip:" + $si + ":" + $sp + ";transport=" + $proto;
In this case all user location records will have the "received" attribut even if a UA isn't behind NAT, but I don't see any problems with that.
This one looks working, but the callee doesn't answer correctly because the TCP isn't correct:
Contact:: sip:528@94.94.X.X:1380;transport=TCP;ob;q=;expires=294;flags=0x0;cflags=0x0;state=0;socket=tcp:178.79.X.X:5060;methods=0x1FDF;received=sip:94.94.X.X:37030;transport=tcp;user_agent=<PJSUA
v2.1-svn Linux-3.10.5.201/x86_64/glibc-2.17>;reg-id=0
The contact uses a port which isn't translated inside by the router, the received field shows the right one. Should I change the Contact header instead? In case, how can I do that?
I didn't understand what "callee doesn't answer correctly" means. Callee doesn't know what's in the received field of its registration.
Sorry! My typo! I meant why kamilio wasn't reusing the TCP port specified in the REGISTER even for the INVITE. I mean, kamailio now knows the received field so I'm expecting it routes the requests for the called UA through the address:port specified in this field.
The only problem I see is that your router changes ip in the contact field of REGISTER requests and then kamailio puts this value (new_ip:old_port) to INVITEs destined to UAs behind your NAT router.
Indeed, my testing router is affected by this ALG "problem"! I guess it's better to disable it and complete simple TCP tests then move to TLS.
It's not likely, but maybe pjsip doesn't like INVITEs with RURI which differs from what it put to the contact during registration. IIRC pjsua prints lots of debugging information. So you can check if pjsua shows anything when INVITE comes in its log and also attach a trace of such a call.
pjsua doesn't report any problem so far, so seems accepting the INVITE.
P.S. IMHO the best way to pass sip through such routers is sip over TLS.
Yep! I'll do that.
Just to say that everything works pretty fine once moved to TLS!
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
-
Daniel-Constantin Mierla -
Roberto Fichera -
Vitaliy Aleksandrov