Hmmm..have you changed the UACs on both side or at least the one that is problematic, like Juha said. <div><br></div><div>In the codec negotiation I see</div><div>a=rtpmap:18 G729/8000 from pjmedia</div><div>and </div><div>
a=rtpmap:18 G729/8000<b>/1</b> from VoIPSip/Switch</div><div><br></div><div>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.</div>
<div><br></div><div>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. <br>
<br><div class="gmail_quote">On Mon, Oct 10, 2011 at 2:18 AM, Austin Einter <span dir="ltr"><<a href="mailto:austin.einter@gmail.com">austin.einter@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">
<div>Hi All</div>
<div>Thanks for your kind answer.</div>
<div> </div>
<div>The call flow looks as below</div>
<div>I have two doubts here</div>
<div> </div>
<div>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</div>
<div> </div>
<div>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.</div>
<div>will this lead to any issues.</div>
<div> </div>
<div>Regards</div>
<div>Austin.</div>
<div><br>INVITE <a href="http://sip:919731573290@134.121.32.130:5060" target="_blank">sip:919731573290@134.121.32.130:5060</a> SIP/2.0</div>
<div>Via: SIP/2.0/UDP 192.168.1.2:53489;rport;branch=z9hG4bKPj0052793130024dda88418cf7a392b7ae</div>
<div>Max-Forwards: 70</div>
<div>From: <a href="mailto:sip%3Aaustin@134.121.32.130" target="_blank">sip:austin@134.121.32.130</a>;tag=8c2e350c064e417c96bda1378470fd46</div>
<div>To: <a href="mailto:sip%3A919731573290@134.121.32.130" target="_blank">sip:919731573290@134.121.32.130</a></div>
<div>Contact: <sip:austin@192.168.1.2:53489;ob></div>
<div>Call-ID: b637fa62393a45a0a58633c1a8f43a86</div>
<div>CSeq: 14417 INVITE</div>
<div>Route: <sip:134.33.8.138:5060;lr></div>
<div>Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS</div>
<div>Supported: replaces, 100rel, timer, norefersub</div>
<div>Session-Expires: 1800</div>
<div>Min-SE: 90</div>
<div>User-Agent: VoIP Client v1.01</div>
<div>Proxy-Authorization: Digest username="austin", realm="VoipSwitch", nonce="131819433109160428210053141040", uri="<a href="http://sip:919731573290@134.121.32.130:5060" target="_blank">sip:919731573290@134.121.32.130:5060</a>", response="935c3130fe07e2413ccf127d5fb6b9d1"</div>
<div>Content-Type: application/sdp</div>
<div>Content-Length: 271</div>
<div> </div>
<div>v=0</div>
<div>o=- <a href="tel:3527202931" value="+13527202931" target="_blank">3527202931</a> <a href="tel:3527202931" value="+13527202931" target="_blank">3527202931</a> IN IP4 192.168.1.2</div>
<div>s=pjmedia</div>
<div>c=IN IP4 192.168.1.2</div>
<div>t=0 0</div>
<div>a=X-nat:0</div>
<div>m=audio 4000 RTP/AVP 18 4 96</div>
<div>a=rtcp:4001 IN IP4 192.168.1.2</div>
<div>a=rtpmap:18 G729/8000</div>
<div>a=rtpmap:4 G723/8000</div>
<div>a=sendrecv</div>
<div>a=rtpmap:96 telephone-event/8000</div>
<div>a=fmtp:96 0-15</div>
<div> </div>
<div>SIP/2.0 100 trying </div>
<div>Via: SIP/2.0/UDP 192.168.1.2:53489;rport=13341;branch=z9hG4bKPj0052793130024dda88418cf7a392b7ae;received=122.178.237.67</div>
<div>From: <a href="mailto:sip%3Aaustin@134.121.32.130" target="_blank">sip:austin@134.121.32.130</a>;tag=8c2e350c064e417c96bda1378470fd46</div>
<div>To: <a href="mailto:sip%3A919731573290@134.121.32.130" target="_blank">sip:919731573290@134.121.32.130</a></div>
<div>Call-ID: b637fa62393a45a0a58633c1a8f43a86</div>
<div>CSeq: 14417 INVITE</div>
<div>Server: kamailio (3.1.5 (i386/linux))</div>
<div>Content-Length: 0</div>
<div> </div>
<div> </div>
<div>SIP/2.0 183 Session Progress</div>
<div>CSeq: 14417 INVITE</div>
<div>Via: SIP/2.0/UDP 192.168.1.2:53489;branch=z9hG4bKPj0052793130024dda88418cf7a392b7ae</div>
<div>From: <a href="mailto:sip%3Aaustin@134.121.32.130" target="_blank">sip:austin@134.121.32.130</a>;tag=8c2e350c064e417c96bda1378470fd46</div>
<div>Call-ID: b637fa62393a45a0a58633c1a8f43a86</div>
<div>To: <a href="mailto:sip%3A919731573290@134.121.32.130" target="_blank">sip:919731573290@134.121.32.130</a>;tag=09100511163117092280006157</div>
<div>Contact: <sip:134.121.32.130:5060;transport=udp></div>
<div>Content-Type: application/sdp</div>
<div>Content-Length: 241</div>
<div>Record-Route: <sip:134.33.8.138;lr=on;nat=yes></div>
<div> </div>
<div>v=0</div>
<div>o=VoipSwitch 6156 7156 IN IP4 134.33.8.138</div>
<div>s=VoipSIP</div>
<div>i=Audio Session</div>
<div>c=IN IP4 134.33.8.138</div>
<div>t=0 0</div>
<div>m=audio 46976 RTP/AVP 18 96</div>
<div>a=rtpmap:18 G729/8000/1</div>
<div>a=rtpmap:96 telephone-event/8000</div>
<div>a=fmtp:96 0-15</div>
<div>a=sendrecv</div>
<div>a=nortpproxy:yes</div>
<div> </div>
<div>SIP/2.0 200 OK</div>
<div>CSeq: 14417 INVITE</div>
<div>Via: SIP/2.0/UDP 192.168.1.2:53489;branch=z9hG4bKPj0052793130024dda88418cf7a392b7ae</div>
<div>From: <a href="mailto:sip%3Aaustin@134.121.32.130" target="_blank">sip:austin@134.121.32.130</a>;tag=8c2e350c064e417c96bda1378470fd46</div>
<div>Call-ID: b637fa62393a45a0a58633c1a8f43a86</div>
<div>To: <a href="mailto:sip%3A919731573290@134.121.32.130" target="_blank">sip:919731573290@134.121.32.130</a>;tag=09100511163117092280006157</div>
<div>Contact: <sip:134.121.32.130:5060;transport=udp></div>
<div>Content-Type: application/sdp</div>
<div>Content-Length: 241</div>
<div>Record-Route: <sip:134.33.8.138;lr=on;nat=yes></div>
<div> </div>
<div>v=0</div>
<div>o=VoipSwitch 6156 7156 IN IP4 134.33.8.138</div>
<div>s=VoipSIP</div>
<div>i=Audio Session</div>
<div>c=IN IP4 134.33.8.138</div>
<div>t=0 0</div>
<div>m=audio 46976 RTP/AVP 18 96</div>
<div>a=rtpmap:18 G729/8000/1</div>
<div>a=rtpmap:96 telephone-event/8000</div>
<div>a=fmtp:96 0-15</div>
<div>a=sendrecv</div>
<div>a=nortpproxy:yes</div>
<div> </div>
<div>ACK sip:134.121.32.130:5060;transport=udp SIP/2.0</div>
<div>Via: SIP/2.0/UDP 192.168.1.2:53489;rport;branch=z9hG4bKPj73092b1de9aa4d4498adac484efacfda</div>
<div>Max-Forwards: 70</div>
<div>From: <a href="mailto:sip%3Aaustin@134.121.32.130" target="_blank">sip:austin@134.121.32.130</a>;tag=8c2e350c064e417c96bda1378470fd46</div>
<div>To: <a href="mailto:sip%3A919731573290@134.121.32.130" target="_blank">sip:919731573290@134.121.32.130</a>;tag=09100511163117092280006157</div>
<div>Call-ID: b637fa62393a45a0a58633c1a8f43a86</div>
<div>CSeq: 14417 ACK</div>
<div>Route: <sip:134.33.8.138;lr;nat=yes></div>
<div>Content-Length: 0</div>
<div> </div>
<div> </div>
<div>BYE sip:austin@122.178.237.67:13341;ob SIP/2.0</div>
<div>Max-Forwards: 10</div>
<div>CSeq: 1 BYE</div>
<div>Via: SIP/2.0/UDP 134.33.8.138;branch=z9hG4bK029.52d62945.0</div>
<div>Via: SIP/2.0/UDP 134.121.32.130:5060;rport=5060;branch=z9hG4bK091005111656091709252938</div>
<div>From: <a href="mailto:sip%3A919731573290@134.121.32.130" target="_blank">sip:919731573290@134.121.32.130</a>;tag=09100511163117092280006157</div>
<div>Call-ID: b637fa62393a45a0a58633c1a8f43a86</div>
<div>To: <a href="mailto:sip%3Aaustin@134.121.32.130" target="_blank">sip:austin@134.121.32.130</a>;tag=8c2e350c064e417c96bda1378470fd46</div>
<div>Content-Length: 0</div>
<div> </div>
<div> </div>
<div>SIP/2.0 200 OK</div>
<div>Via: SIP/2.0/UDP 134.33.8.138;received=134.33.8.138;branch=z9hG4bK029.52d62945.0</div>
<div>Via: SIP/2.0/UDP 134.121.32.130:5060;rport=5060;branch=z9hG4bK091005111656091709252938</div>
<div>Call-ID: b637fa62393a45a0a58633c1a8f43a86</div>
<div>From: <<a href="mailto:sip%3A919731573290@134.121.32.130" target="_blank">sip:919731573290@134.121.32.130</a>>;tag=09100511163117092280006157</div>
<div>To: <<a href="mailto:sip%3Aaustin@134.121.32.130" target="_blank">sip:austin@134.121.32.130</a>>;tag=8c2e350c064e417c96bda1378470fd46</div>
<div>CSeq: 1 BYE</div>
<div>Content-Length: 0</div><div><div></div><div class="h5">
<div> </div>
<div><br><br> </div>
<div class="gmail_quote">On Sun, Oct 9, 2011 at 11:50 AM, Sammy Govind <span dir="ltr"><<a href="mailto:govoiper@gmail.com" target="_blank">govoiper@gmail.com</a>></span> wrote:<br>
<blockquote style="border-left:#ccc 1px solid;margin:0px 0px 0px 0.8ex;padding-left:1ex" class="gmail_quote">Hey,
<div>Can you send in the SIP/SDP invites. I suspect the codecs issue here.</div>
<div>--</div>
<div>Regards,</div>
<div>Sammy<br>
<div><br><br>
<div class="gmail_quote">
<div>
<div></div>
<div>On Sun, Oct 9, 2011 at 8:57 AM, Austin Einter <span dir="ltr"><<a href="mailto:austin.einter@gmail.com" target="_blank">austin.einter@gmail.com</a>></span> wrote:<br></div></div>
<blockquote style="border-left:#ccc 1px solid;margin:0px 0px 0px 0.8ex;padding-left:1ex" class="gmail_quote">
<div>
<div></div>
<div>
<div>Hi </div>
<div>I am using Kamailio 3.1.5 . I am using RTP proxy also.</div>
<div>I have used default kamailio.cfg.sample fiile , and just added line #!define WITH_NAT.</div>
<div> </div>
<div>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.</div>
<div> </div>
<div>With this I found, call is established, all signalling and media packets are passing through kamailio / rtp-proxy.</div>
<div>So far so good.</div>
<div> </div>
<div>One way audio stream (from called party to calling party) quality is good.</div>
<div>The other audio stream (from calling party to called party is very bad.</div>
<div> </div>
<div>Did anybody face this issue? Please help me to sort out this issue audio quality issue.</div>
<div> </div>
<div>Regards</div>
<div>Austin</div><font color="#888888">
<div> </div>
<div> </div></font><br></div></div>
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SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list<br>
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