<div dir="ltr"><div><br></div><div><div>Hi,</div><div><br></div><div>Thanks very much for this, that solved the double-m-line issue. Now I'm calling rtpengine_offer in a branch route.</div><div><br></div><div>One issue still remains; the call still gets connected to the called zoiper client, but it gets hung up right away. I traced this to be caused by a BYE message from Kamailio, which I think may be caused by the fact that the SDP returning to the chrome/websocket caller contains RTP/AVP profile, which it doesn't support. What I don't know is why this happens. </div>
<div><br></div><div>I have a reply route: </div><div><br></div><div>onreply_route[RTPPROXY_REPLY] {</div><div> if (status =~ "18[03]") {</div><div> # early media won't work so strip it out now to avoid problems</div>
<div>xlog("L_INFO","RTPPROXY_REPLY: Changing 18[03] to 180 Ringing and removing sdp body");</div><div> change_reply_status(180, "Ringing");</div><div> remove_body();</div>
<div> } else if (status =~ "2[0-9][0-9]" && sdp_content()) {</div><div>xlog("L_INFO","Got status 2xx, calling rtpengine_answer with $avp(rtpproxy_answer_flags)");</div><div> rtpengine_answer($avp(rtpproxy_answer_flags));</div>
<div> }</div><div>}</div><div><br></div><div><br></div><div>I suspect there is a 200 OK message going from Kamailio to the ws client that has the RTP/AVP profile, as the Jssip library gets an error stating: "Failed to set remote answer sdp: Failed to push down transport description: Answerer must use either active or passive value for setup attribute."</div>
<div><br></div><div>Any idea on what's going wrong here? </div><div><br></div><div><br></div><div>I'm not sure how to fix my config, maybe this log can help; this is printed to log as the called zoiper client answers the call:</div>
<div><br></div><div>Note: </div><div>at 'Jul 22 19:36:36 u363id562 kamailio[27122]' I get the output from above route block</div><div>the next output at 'Jul 22 19:36:36 u363id562 rtpengine[16930]' is somewhat confusing to me, it still mentions RTP/AVP and later RTP/SAVPF.</div>
<div>Again here 1.1.1.1 is the public ip of my Kamailio & Asterisk machine and 2.2.2.2 is the ip behind which my clients reside.</div><div><br></div><div><br></div><div>Jul 22 19:36:36 u363id562 kamailio[27121]: NOTICE: acc [acc.c:315]: acc_log_request(): ACC: transaction answered: timestamp=1406043396;method=INVITE;from_tag=as7d3db027;to_tag=696faf28;call_id=<a href="mailto:5c066eb83cb7a0e144248fc344ead8d1@testers.com">5c066eb83cb7a0e144248fc344ead8d1@testers.com</a>;code=200;reason=OK;src_user=771;src_domain=<a href="http://testers.com">testers.com</a>;src_ip=1.1.1.1;dst_ouser=771;dst_user=771;dst_domain=2.2.2.2</div>
<div>Jul 22 19:36:36 u363id562 kamailio[27123]: INFO: <script>: WITHINDLG: hooked to MANAGE_UA_FAILURE route</div><div>Jul 22 19:36:36 u363id562 kamailio[27122]: INFO: <script>: RTPPROXY_REPLY: status 2xx, calling rtpengine_answer with rtcp-mux-demux trust-address replace-origin replace-session-connection ICE=force RTP/SAVPF</div>
<div>Jul 22 19:36:36 u363id562 rtpengine[16930]: Got valid command from <a href="http://127.0.0.1:32955">127.0.0.1:32955</a>: answer - { "sdp": "v=0#015#012o=root 1988031334 1988031334 IN IP4 1.1.1.1#015#012s=Asterisk PBX 11.10.2#015#012c=IN IP4 1.1.1.1#015#012t=0 0#015#012m=audio 16560 RTP/AVP 0 8 126#015#012a=rtpmap:0 PCMU/8000#015#012a=rtpmap:8 PCMA/8000#015#012a=rtpmap:126 telephone-event/8000#015#012a=fmtp:126 0-16#015#012a=ptime:20#015#012a=ice-ufrag:4e7625e7134e85a032af6b6a68090b1d#015#012a=ice-pwd:78db4d825604911059fa381667e9ec4c#015#012a=candidate:H5bdd423d 1 UDP 2130706431 1.1.1.1 16560 typ host#015#012a=candidate:H5bdd423d 2 UDP 2130706430 1.1.1.1 16561 typ host#015#012a=sendrecv#015#012", "ICE": "force", "flags": [ "trust-address" ], "replace": [ "origin", "session-connection" ], "transport-protocol": "RTP/SAVPF", "rtcp-mux": [ "demux" ], "call-id": "35hkg7ct44fpndrs0gjt", "received-from": [ "IP4", "1.1.1.1" ], "from-tag": "ulaa4af3s4", "to-tag": "as6bed69a3", "command": "answer" }</div>
<div>Jul 22 19:36:36 u363id562 rtpengine[16930]: [35hkg7ct44fpndrs0gjt] Returning to SIP proxy: d3:sdp535:v=0#015#012o=root 1988031334 1988031334 IN IP4 1.1.1.1#015#012s=Asterisk PBX 11.10.2#015#012c=IN IP4 1.1.1.1#015#012t=0 0#015#012a=ice-lite#015#012m=audio 30150 RTP/SAVPF 0 8 126#015#012a=rtpmap:0 PCMU/8000#015#012a=rtpmap:8 PCMA/8000#015#012a=rtpmap:126 telephone-event/8000#015#012a=fmtp:126 0-16#015#012a=ptime:20#015#012a=sendrecv#015#012a=rtcp:30150#015#012a=rtcp-mux#015#012a=setup:actpass#015#012a=fingerprint:sha-1 60:14:ED:9B:C6:80:D2:00:F9:90:46:E9:26:CF:AB:D7:23:03:B5:11#015#012a=ice-ufrag:12gFxwCc#015#012a=ice-pwd:YWrFa0E6PgUMIDoFQzZSaCRbE5O9#015#012a=candidate:aPQVJN7QRnK926OO 1 UDP 2130706431 1.1.1.1 30150 typ host#015#0126:result2:oke</div>
<div>Jul 22 19:36:36 u363id562 kamailio[27122]: NOTICE: acc [acc.c:315]: acc_log_request(): ACC: transaction answered: timestamp=1406043396;method=INVITE;from_tag=ulaa4af3s4;to_tag=as6bed69a3;call_id=35hkg7ct44fpndrs0gjt;code=200;reason=OK;src_user=660;src_domain=<a href="http://testers.com">testers.com</a>;src_ip=2.2.2.2;dst_ouser=771;dst_user=771;dst_domain=<a href="http://testers.com">testers.com</a></div>
<div>Jul 22 19:36:36 u363id562 kamailio[27151]: INFO: <script>: WITHINDLG: hooked to MANAGE_UA_FAILURE route</div><div>Jul 22 19:36:36 u363id562 kamailio[27151]: INFO: <script>: WITHINDLG: hooked to MANAGE_UA_FAILURE route</div>
<div>Jul 22 19:36:36 u363id562 rtpengine[16930]: Got valid command from <a href="http://127.0.0.1:39090">127.0.0.1:39090</a>: delete - { "call-id": "35hkg7ct44fpndrs0gjt", "received-from": [ "IP4", "2.2.2.2" ], "from-tag": "ulaa4af3s4", "command": "delete" }</div>
<div>Jul 22 19:36:36 u363id562 rtpengine[16930]: [35hkg7ct44fpndrs0gjt] Deleting full call</div><div>Jul 22 19:36:36 u363id562 rtpengine[16930]: [35hkg7ct44fpndrs0gjt] Final packet stats:</div><div>Jul 22 19:36:36 u363id562 rtpengine[16930]: [35hkg7ct44fpndrs0gjt] --- Tag 'as6bed69a3', created 0:05 ago, in dialogue with 'ulaa4af3s4'</div>
<div>Jul 22 19:36:36 u363id562 rtpengine[16930]: [35hkg7ct44fpndrs0gjt] ------ Media #1, port 30150 <> <a href="http://2.2.2.2:56050">2.2.2.2:56050</a>, 0 p, 0 b, 0 e</div><div>Jul 22 19:36:36 u363id562 rtpengine[16930]: [35hkg7ct44fpndrs0gjt] --- Tag 'ulaa4af3s4', created 0:05 ago, in dialogue with 'as6bed69a3'</div>
<div>Jul 22 19:36:36 u363id562 rtpengine[16930]: [35hkg7ct44fpndrs0gjt] ------ Media #1, port 30148 <> <a href="http://1.1.1.1:16560">1.1.1.1:16560</a>, 0 p, 0 b, 0 e</div><div>Jul 22 19:36:36 u363id562 rtpengine[16930]: [35hkg7ct44fpndrs0gjt] ------ Media #1, port 30149 <> <a href="http://1.1.1.1:16561">1.1.1.1:16561</a> (RTCP), 0 p, 0 b, 0 e</div>
<div>Jul 22 19:36:36 u363id562 rtpengine[16930]: [35hkg7ct44fpndrs0gjt] Returning to SIP proxy: d7:createdi1406043391e11:last signali1406043396e4:tagsd10:ulaa4af3s4d3:tag10:ulaa4af3s47:createdi1406043391e16:in dialogue with10:as6bed69a36:mediasld5:indexi1e4:type5:audio8:protocol7:RTP/AVP7:streamsld10:local porti30148e8:endpointd6:family4:IPv47:address12:1.1.1.14:porti16560ee19:advertised endpointd6:family4:IPv47:address12:1.1.1.14:porti16560ee11:last packeti1406043391e5:flagsl3:RTP6:fillede5:statsd7:packetsi0e5:bytesi0e6:errorsi0eeed10:local porti30149e8:endpointd6:family4:IPv47:address12:1.1.1.14:porti16561ee19:advertised endpointd6:family4:IPv47:address12:1.1.1.14:porti16561ee11:last packeti1406043391e5:flagsl4:RTCP6:fillede5:statsd7:packetsi0e5:bytesi0e6:errorsi0eeee5:flagsl11:initialized3:ICEeeee10:as6bed69a3d3:tag10:as6bed69a37:createdi1406043391e16:in dialogue with10:ulaa4af3s46:mediasld5:indexi1e4:type5:audio8:protocol9:RTP/SAVPF7:streamsld10:local porti30150e8:endpointd6:family4:IPv47:address12:2.2.2.24:porti56050ee19:advertised endpointd6:family4:IPv47:address12:2.2.2.24:porti56050ee11:last packeti1406043391e5:flagsl3:RTP4:RTCP6:fillede5:statsd7:packetsi0e5:bytesi0e6:errorsi0eeed10:local porti30151e8:endpointd6:family4:IPv67:address2:::4:porti0ee19:advertised endpointd6:family4:IPv67:address2:::4:porti0ee11:last packeti1406043391e5:flagsl4:RTCP13:fallback RTCP6:fillede5:statsd7:packetsi0e5:bytesi0e6:errorsi0eeee5:flagsl11:initialized8:rtcp-mux9:DTLS-SRTP3:ICEeeeee6:totalsd3:RTPd7:packetsi0e5:bytesi0e6:errorsi0ee4:RTCPd7:packetsi0e5:bytesi0e6:errorsi0eee6:result2:oke</div>
<div>Jul 22 19:36:36 u363id562 kamailio[27124]: NOTICE: acc [acc.c:315]: acc_log_request(): ACC: transaction answered: timestamp=1406043396;method=BYE;from_tag=ulaa4af3s4;to_tag=as6bed69a3;call_id=35hkg7ct44fpndrs0gjt;code=200;reason=OK;src_user=660;src_domain=<a href="http://testers.com">testers.com</a>;src_ip=2.2.2.2;dst_ouser=771;dst_user=771;dst_domain=1.1.1.1</div>
<div>Jul 22 19:36:36 u363id562 kamailio[27121]: INFO: <script>: WITHINDLG: hooked to MANAGE_UA_FAILURE route</div><div>Jul 22 19:36:36 u363id562 kamailio[27122]: NOTICE: acc [acc.c:315]: acc_log_request(): ACC: transaction answered: timestamp=1406043396;method=BYE;from_tag=as7d3db027;to_tag=696faf28;call_id=<a href="mailto:5c066eb83cb7a0e144248fc344ead8d1@testers.com">5c066eb83cb7a0e144248fc344ead8d1@testers.com</a>;code=200;reason=OK;src_user=771;src_domain=<a href="http://testers.com">testers.com</a>;src_ip=1.1.1.1;dst_ouser=771;dst_user=771;dst_domain=2.2.2.2</div>
<div><br></div><div><br></div><div><br></div><div>cheers,</div><div>Olli</div></div><div><br></div><div><br></div></div><div class="gmail_extra"><br><br><div class="gmail_quote">2014-07-21 16:38 GMT+03:00 Richard Fuchs <span dir="ltr"><<a href="mailto:rfuchs@sipwise.com" target="_blank">rfuchs@sipwise.com</a>></span>:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">On 20/07/14 01:15 PM, Olli Heiskanen wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
Hi,<br>
</blockquote>
...<div class=""><br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
There may be something off in my Asterisk configs since it's Asterisk<br>
that responds 488, but see how Kamailio responds, SDP contains 2 similar<br>
m= lines. Is there something I might be doing wrong in configuring<br>
rtpengine? The INVITE going to the called client has nice clean rtp with<br>
RTP/AVP profile.<br>
</blockquote>
<br></div>
This looks a lot like the issue discussed here:<br>
<br>
<a href="http://lists.sip-router.org/pipermail/sr-dev/2014-July/024507.html" target="_blank">http://lists.sip-router.org/<u></u>pipermail/sr-dev/2014-July/<u></u>024507.html</a><br>
<br>
The solution was:<br>
<br>
<a href="http://lists.sip-router.org/pipermail/sr-dev/2014-July/024519.html" target="_blank">http://lists.sip-router.org/<u></u>pipermail/sr-dev/2014-July/<u></u>024519.html</a><div class="HOEnZb"><div class="h5"><br>
<br>
cheers<br>
<br>
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</div></div></blockquote></div><br></div>