Hi All Thanks for your kind answer. The call flow looks as below I have two doubts here
1. 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 2. 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 <govoi...@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.ein...@gmail.com>wrote: > >> 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 > >
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