Hmmm..have you changed the UACs on both side or at least the one that is problematic, like Juha said.
In the codec negotiation I see a=rtpmap:18 G729/8000 from pjmedia and a=rtpmap:18 G729/8000*/1* from VoIPSip/Switch
I'm not sure that this channel info plays much role here because according to rfc it could be omitted if its 1 and no extra params are in the string. But I don't know if this could cause the other UAC to behave abnormally.
you can also confirm if this is Server side issue or UAC side issue by taking a full size tcpdump on ther server for this particular call and hear the call using wireshark. A faulty client side behaviour can be identified if the audio on both sides is ok on server.
On Mon, Oct 10, 2011 at 2:18 AM, Austin Einter austin.einter@gmail.comwrote:
Hi All Thanks for your kind answer.
The call flow looks as below I have two doubts here
- My UA is just behind the Modem, and in Kamailio config file I have
enabled WITH_NAT, will this lead to any kind of problem
- In kamailio proxy I am using force_rtp_proxy and unforce_rtp_proxy
instead of rtpproxy_offer/rtpproxy_answer. Not sure whats the corresponding api for unforce_rtp_proxy. will this lead to any issues.
Regards Austin.
INVITE sip:919731573290@134.121.32.130:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.2:53489 ;rport;branch=z9hG4bKPj0052793130024dda88418cf7a392b7ae Max-Forwards: 70 From: sip:austin@134.121.32.130;tag=8c2e350c064e417c96bda1378470fd46 To: sip:919731573290@134.121.32.130 Contact: sip:austin@192.168.1.2:53489;ob Call-ID: b637fa62393a45a0a58633c1a8f43a86 CSeq: 14417 INVITE Route: sip:134.33.8.138:5060;lr Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Supported: replaces, 100rel, timer, norefersub Session-Expires: 1800 Min-SE: 90 User-Agent: VoIP Client v1.01 Proxy-Authorization: Digest username="austin", realm="VoipSwitch", nonce="131819433109160428210053141040", uri=" sip:919731573290@134.121.32.130:5060", response="935c3130fe07e2413ccf127d5fb6b9d1" Content-Type: application/sdp Content-Length: 271
v=0 o=- 3527202931 3527202931 IN IP4 192.168.1.2 s=pjmedia c=IN IP4 192.168.1.2 t=0 0 a=X-nat:0 m=audio 4000 RTP/AVP 18 4 96 a=rtcp:4001 IN IP4 192.168.1.2 a=rtpmap:18 G729/8000 a=rtpmap:4 G723/8000 a=sendrecv a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-15
SIP/2.0 100 trying Via: SIP/2.0/UDP 192.168.1.2:53489 ;rport=13341;branch=z9hG4bKPj0052793130024dda88418cf7a392b7ae;received=122.178.237.67 From: sip:austin@134.121.32.130;tag=8c2e350c064e417c96bda1378470fd46 To: sip:919731573290@134.121.32.130 Call-ID: b637fa62393a45a0a58633c1a8f43a86 CSeq: 14417 INVITE Server: kamailio (3.1.5 (i386/linux)) Content-Length: 0
SIP/2.0 183 Session Progress CSeq: 14417 INVITE Via: SIP/2.0/UDP 192.168.1.2:53489 ;branch=z9hG4bKPj0052793130024dda88418cf7a392b7ae From: sip:austin@134.121.32.130;tag=8c2e350c064e417c96bda1378470fd46 Call-ID: b637fa62393a45a0a58633c1a8f43a86 To: sip:919731573290@134.121.32.130;tag=09100511163117092280006157 Contact: sip:134.121.32.130:5060;transport=udp Content-Type: application/sdp Content-Length: 241 Record-Route: sip:134.33.8.138;lr=on;nat=yes
v=0 o=VoipSwitch 6156 7156 IN IP4 134.33.8.138 s=VoipSIP i=Audio Session c=IN IP4 134.33.8.138 t=0 0 m=audio 46976 RTP/AVP 18 96 a=rtpmap:18 G729/8000/1 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-15 a=sendrecv a=nortpproxy:yes
SIP/2.0 200 OK CSeq: 14417 INVITE Via: SIP/2.0/UDP 192.168.1.2:53489 ;branch=z9hG4bKPj0052793130024dda88418cf7a392b7ae From: sip:austin@134.121.32.130;tag=8c2e350c064e417c96bda1378470fd46 Call-ID: b637fa62393a45a0a58633c1a8f43a86 To: sip:919731573290@134.121.32.130;tag=09100511163117092280006157 Contact: sip:134.121.32.130:5060;transport=udp Content-Type: application/sdp Content-Length: 241 Record-Route: sip:134.33.8.138;lr=on;nat=yes
v=0 o=VoipSwitch 6156 7156 IN IP4 134.33.8.138 s=VoipSIP i=Audio Session c=IN IP4 134.33.8.138 t=0 0 m=audio 46976 RTP/AVP 18 96 a=rtpmap:18 G729/8000/1 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-15 a=sendrecv a=nortpproxy:yes
ACK sip:134.121.32.130:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.1.2:53489 ;rport;branch=z9hG4bKPj73092b1de9aa4d4498adac484efacfda Max-Forwards: 70 From: sip:austin@134.121.32.130;tag=8c2e350c064e417c96bda1378470fd46 To: sip:919731573290@134.121.32.130;tag=09100511163117092280006157 Call-ID: b637fa62393a45a0a58633c1a8f43a86 CSeq: 14417 ACK Route: sip:134.33.8.138;lr;nat=yes Content-Length: 0
BYE sip:austin@122.178.237.67:13341;ob SIP/2.0 Max-Forwards: 10 CSeq: 1 BYE Via: SIP/2.0/UDP 134.33.8.138;branch=z9hG4bK029.52d62945.0 Via: SIP/2.0/UDP 134.121.32.130:5060 ;rport=5060;branch=z9hG4bK091005111656091709252938 From: sip:919731573290@134.121.32.130;tag=09100511163117092280006157 Call-ID: b637fa62393a45a0a58633c1a8f43a86 To: sip:austin@134.121.32.130;tag=8c2e350c064e417c96bda1378470fd46 Content-Length: 0
SIP/2.0 200 OK Via: SIP/2.0/UDP 134.33.8.138;received=134.33.8.138;branch=z9hG4bK029.52d62945.0 Via: SIP/2.0/UDP 134.121.32.130:5060 ;rport=5060;branch=z9hG4bK091005111656091709252938 Call-ID: b637fa62393a45a0a58633c1a8f43a86 From: sip:919731573290@134.121.32.130;tag=09100511163117092280006157 To: sip:austin@134.121.32.130;tag=8c2e350c064e417c96bda1378470fd46 CSeq: 1 BYE Content-Length: 0
On Sun, Oct 9, 2011 at 11:50 AM, Sammy Govind govoiper@gmail.com wrote:
Hey, Can you send in the SIP/SDP invites. I suspect the codecs issue here. -- Regards, Sammy
On Sun, Oct 9, 2011 at 8:57 AM, Austin Einter austin.einter@gmail.comwrote:
Hi I am using Kamailio 3.1.5 . I am using RTP proxy also. I have used default kamailio.cfg.sample fiile , and just added line #!define WITH_NAT.
I have another Main proxy. I wanted all my signalling and media packets should just pass through machine where Kamailio and RTP proxy are running.
With this I found, call is established, all signalling and media packets are passing through kamailio / rtp-proxy. So far so good.
One way audio stream (from called party to calling party) quality is good. The other audio stream (from calling party to called party is very bad.
Did anybody face this issue? Please help me to sort out this issue audio quality issue.
Regards Austin
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