[SR-Users] Problem Proxy

Nicolas Breuer Nicolas.Breuer at belcenter.biz
Wed Jan 24 13:00:44 CET 2018


Hello Daniel,

I removed the « r » flag
No sound on incoming call after that.

SDP received from the targeted UA
v=0.
o=UserA 198677311 3791273416 IN IP4 192.168.1.97.
s=Session SDP.
c=IN IP4 192.168.1.97.
t=0 0.
m=audio 49152 RTP/AVP 9 101.
a=rtpmap:9 G722/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.


Proxy debug
Jan 24 12:49:55 proxy2 rtpproxy[1291]: INFO:rxmit_packets:1711966639_66898263 at 84.14.241.179: callee's address latched in: 217.112.180.70:49152 (RTP)
Jan 24 12:49:58 proxy2 rtpproxy[1291]: INFO:rxmit_packets:1711966639_66898263 at 84.14.241.179: callee's address latched in: 217.112.180.70:49153 (RTCP)

The port 49152 is the correct port – I received the two streams from each end.

UA 1 : U 2018/01/24 12:51:29.952975 217.112.180.70:49152 -> 217.112.180.9:12096
UA 2 : U 2018/01/24 12:59:34.712932 84.14.241.175:5850 -> 217.112.180.9:12646

But seems there are not bridged ?
Works well with “r” flag






De : Daniel-Constantin Mierla [mailto:miconda at gmail.com]
Envoyé : mercredi 24 janvier 2018 09:49
À : Kamailio (SER) - Users Mailing List <sr-users at lists.kamailio.org>; Nicolas Breuer <Nicolas.Breuer at belcenter.biz>
Objet : Re: [SR-Users] Problem Proxy


Hello,

you have to remove the 'r' flag, it instructs rtpproxy to trust the ip address from sdp:

  - https://www.kamailio.org/docs/modules/stable/modules/rtpproxy.html#rtpproxy.f.rtpproxy_ofrer

Cheers,
Daniel

On 24.01.18 09:27, Nicolas Breuer wrote:
Hello,

Anyone knows how the rtpproxy get the public IP Address in case a private IP is in SDP
The flags of the proxy are “corbf”

If I know how the proxy get this information, I can compare the two SIP frames when bridging the call.


De : sr-users [mailto:sr-users-bounces at lists.kamailio.org] De la part de Dave & Hazel
Envoyé : lundi 22 janvier 2018 13:35
À : 'Kamailio (SER) - Users Mailing List' <sr-users at lists.kamailio.org><mailto:sr-users at lists.kamailio.org>
Objet : Re: [SR-Users] Problem Proxy

Exactly,

It appears that the destination of the 1st call (217.112.180.235) is NAT aware but the 2nd (79.137.49.139) isn’t. The latter is probably using the local IP in the SDP rather than the public IP detectable by examining the IP of the incoming RTP.

What’s the difference between these 2 UAs?


From: sr-users [mailto:sr-users-bounces at lists.kamailio.org] On Behalf Of David Villasmil
Sent: Monday, January 22, 2018 11:42 AM
To: Kamailio (SER) - Users Mailing List <sr-users at lists.kamailio.org<mailto:sr-users at lists.kamailio.org>>
Subject: Re: [SR-Users] Problem Proxy


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<mailto:Nicolas.Breuer at belcenter.biz>> wrote:
Hello,

Any ideas on this ? 😊


De : sr-users [mailto:sr-users-bounces at lists.kamailio.org<mailto:sr-users-bounces at lists.kamailio.org>] De la part de Nicolas Breuer
Envoyé : jeudi 18 janvier 2018 23:49
À : Kamailio (SER) - Users Mailing List <sr-users at lists.kamailio.org<mailto: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<http://217.112.180.235:1024> -> 217.112.180.10:5060<http://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@ <sip:32XXXXXX87@%20> >;tag=gK0d4dfe6f.
Call-ID: 940401290_111374574 at 84.14.241.179<mailto: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 don’t 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<http://217.112.180.235:49155> (RTCP)

Don’t know why because the only information in SDP is 192.168.2.2
The Kamailio didn’t 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<http://217.112.180.235:1024> -> 217.112.180.10:5060<http://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@ <sip:32XXXXXX87@%20> >;tag=gK0b5d7652.
Call-ID: 940248136_13287504 at 84.14.241.179<mailto: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 “192.168.2.2” and this is the reason of the “no audio”.


Jan 18 21:51:26 proxy1 rtpproxy[1314]: INFO:rtpp_command_ul_handle:940248136_13287504 at 84.14.241.179<mailto:INFO%3Artpp_command_ul_handle%3A940248136_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<mailto:INFO%3Artpp_command_ul_handle%3A940248136_13287504 at 84.14.241.179>: pre-filling callee's address with 192.168.2.2:49156<http://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<http://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<http://217.112.180.235:49155> (RTCP)
Jan 18 21:51:32 proxy1 rtpproxy[1314]: INFO:rxmit_packets:940248136_13287504 at 84.14.241.179<mailto:INFO%3Arxmit_packets%3A940248136_13287504 at 84.14.241.179>: callee's address filled in: 217.112.180.235:49157<http://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,


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