[SR-Users] Problem Proxy
David Villasmil
david.villasmil.work at gmail.com
Mon Jan 22 12:42:05 CET 2018
Hello
Without looking at the config. It looks like you're not doing nat
traversal. Are you using a default config? Which one?
On Mon, Jan 22, 2018, 09:21 Nicolas Breuer <Nicolas.Breuer at belcenter.biz>
wrote:
> Hello,
>
>
>
> Any ideas on this ? đ
>
>
>
>
>
> *De :* sr-users [mailto:sr-users-bounces at lists.kamailio.org] *De la part
> de* Nicolas Breuer
> *Envoyé :* jeudi 18 janvier 2018 23:49
> *Ă :* Kamailio (SER) - Users Mailing List <sr-users at lists.kamailio.org>
> *Objet :* [SR-Users] Problem Proxy
>
>
>
>
>
> Hello the list,
>
>
>
> I have a problem on the proxy with the audio between two calls bridged by
> a UAC.
>
> When I made a normal call, no problems.
>
>
>
> My UAC is nated.
>
> UAC > Router > KAMAILIO
>
>
>
> Frames arrives with private IP in the SDP.
>
>
>
> U 2018/01/18 21:50:16.798581 217.112.180.235:1024 -> 217.112.180.10:5060
>
> SIP/2.0 200 OK.
>
> Via: SIP/2.0/UDP
> 217.112.180.10;branch=z9hG4bK2d06.2f2a1856bde881173a3fc413c4136b83.0.
>
> Via: SIP/2.0/UDP 84.14.241.179:5060
> ;rport=5060;branch=z9hG4bK0dBe3143bcf7f60a70b.
>
> Record-Route: <
> sip:217.112.180.10;lr=on;ftag=gK0d4dfe6f;did=227.c0d2;nat=yes>.
>
> From: <sip:32XXXXXX87@ >;tag=gK0d4dfe6f.
>
> Call-ID: 940401290_111374574 at 84.14.241.179.
>
> CSeq: 29328 INVITE.
>
> Contact: <sip:32XXXXXX61 at 192.168.2.2:5060;transport=udp>.
>
> Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,INFO,NOTIFY.
>
> Supported: timer,100rel.
>
> Server: IP Office 10.1.0.0.0 build 237.
>
> Min-SE: 1800.
>
> Require: timer.
>
> Session-Expires: 1800;refresher=uas.
>
> To: <sip:32XXXXXX61 at 217.112.180.235>;tag=998e429e819ba686.
>
> Content-Type: application/sdp.
>
> Content-Length: 202.
>
> .
>
> v=0.
>
> o=UserA 3721571424 1025404311 IN IP4 192.168.2.2.
>
> s=Session SDP.
>
> c=IN IP4 192.168.2.2.
>
> t=0 0.
>
> m=audio 49154 RTP/AVP 9 101.
>
> a=rtpmap:9 G722/8000.
>
> a=rtpmap:101 telephone-event/8000.
>
> a=fmtp:101 0-15.
>
>
>
> I donât understand why but the proxy (in case of an incoming call) succeed
> to determine the public IP.
>
>
>
> *Jan 18 21:40:08 proxy1 rtpproxy[1314]:
> INFO:rxmit_packets:1fb3f1c6c8dfe738221842406be48450: caller's address
> filled in: 217.112.180.235:49154 <http://217.112.180.235:49154> (RTP)*
>
> Jan 18 21:40:08 proxy1 rtpproxy[1314]:
> INFO:rxmit_packets:1fb3f1c6c8dfe738221842406be48450: guessing RTCP port for
> caller to be 49155
>
> Jan 18 21:40:16 proxy1 rtpproxy[1314]:
> INFO:rxmit_packets:1fb3f1c6c8dfe738221842406be48450: caller's address
> latched in: 217.112.180.235:49155 (RTCP)
>
>
>
> Donât know why because the only information in SDP is 192.168.2.2
>
> The Kamailio didnât send the information of the proxy to the UAC , but to
> the other end as this is an incoming call.
>
>
>
>
>
> So I have audio in this case.
>
>
>
> When I setup a bridge on the UAC to a second number, we have an issue. (
> Kamailio 4.4.6 )
>
>
>
> This is the same frames
>
>
>
> U 2018/01/18 21:51:26.607270 217.112.180.235:1024 -> 217.112.180.10:5060
>
> SIP/2.0 200 OK.
>
> Via: SIP/2.0/UDP
> 217.112.180.10;branch=z9hG4bK4181.bb7aab84b6fb0aea5836b4d8874406ec.0.
>
> Via: SIP/2.0/UDP 84.14.241.179:5060
> ;rport=5060;branch=z9hG4bK0bB16ee5a3d300fee16.
>
> Record-Route: <
> sip:217.112.180.10;lr=on;ftag=gK0b5d7652;did=cc6.5452;nat=yes>.
>
> From: <sip:32XXXXXX87@ >;tag=gK0b5d7652.
>
> Call-ID: 940248136_13287504 at 84.14.241.179.
>
> CSeq: 21946 INVITE.
>
> Contact: <sip:32XXXXXX61 at 192.168.2.2:5060;transport=udp>.
>
> Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,INFO,NOTIFY.
>
> Supported: timer,100rel.
>
> Server: IP Office 10.1.0.0.0 build 237.
>
> Min-SE: 1800.
>
> Require: timer.
>
> Session-Expires: 1800;refresher=uas.
>
> To: <sip:32XXXXXX61 at 217.112.180.235>;tag=559c99f5edcab5d4.
>
> Content-Type: application/sdp.
>
> Content-Length: 202.
>
> .
>
> v=0.
>
> o=UserA 1781830446 4071482272 IN IP4 192.168.2.2.
>
> s=Session SDP.
>
> c=IN IP4 192.168.2.2.
>
> t=0 0.
>
> m=audio 49156 RTP/AVP 8 101.
>
> a=rtpmap:8 PCMA/8000.
>
> a=rtpmap:101 telephone-event/8000.
>
> a=fmtp:101 0-16.
>
>
>
> But in this case no audio
>
>
>
> RTCP detected but no the RTP.
>
> He took the private ip address â192.168.2.2â and this is the reason of the
> âno audioâ.
>
>
>
>
>
> Jan 18 21:51:26 proxy1 rtpproxy[1314]:
> INFO:rtpp_command_ul_handle:940248136_13287504 at 84.14.241.179: lookup on
> ports 11264/10446, session timer restarted
>
> Jan 18 21:51:26 proxy1 rtpproxy[1314]:
> INFO:rtpp_command_ul_handle:940248136_13287504 at 84.14.241.179: pre-filling
> callee's address with 192.168.2.2:49156
>
> Jan 18 21:51:26 proxy1 rtpproxy[1314]:
> INFO:rxmit_packets:e1e387824e0753522b7bc55e02090784: callee's address
> latched in: 79.137.49.139:39176 (RTP)
>
> Jan 18 21:51:32 proxy1 rtpproxy[1314]:
> INFO:rxmit_packets:e1e387824e0753522b7bc55e02090784: caller's address
> filled in: 217.112.180.235:49155 (RTCP)
>
> Jan 18 21:51:32 proxy1 rtpproxy[1314]:
> INFO:rxmit_packets:940248136_13287504 at 84.14.241.179: callee's address
> filled in: 217.112.180.235:49157 (RTCP)
>
>
>
> I would like to understand why with the first call, no issues to determine
> the RTP IP and not in the second case,
>
>
>
>
> _______________________________________________
> Kamailio (SER) - Users Mailing List
> sr-users at lists.kamailio.org
> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.kamailio.org/pipermail/sr-users/attachments/20180122/68aa008e/attachment.html>
More information about the sr-users
mailing list