Hello, Thanks Gareth for your reply. I assume you're referring to the first INVITE in my message, which is from the call that works. I don't know why the sdp displays only iLBC and speex at that point but the Zoiper client that's making the call is configured to support gsm, speex, ulaw, alaw, and iLBC, and the call works fine, audio and all, as the sdp that leaves Asterisk (thus reaches the called peer) actually contains ulaw, gsm and alaw.
In the failing case Asterisk sends the INVITE via Kamailio to the called webrtc client, and in this message the rtp profile is m=audio 12902 RTP/AVP 0 3 8 101. Kamailio sends the INVITE to the client, which responds with 488. Kamailio notices this and uses rtpengine to handle the rtp, but: the client will not accept a second INVITE even though the sdp is correct this time: the client responds with 482 Loop Detected because the Call-ID is the same as the previous INVITE it got. This is why I can't handle the rtp using rtpengine, and here things have already gone wrong. So I need the INVITE to contain correct sdp when it leaves Asterisk, so sdp conversion and rtpengine would net be needed. Wonder if there's any way to do that? cheers, Olli 2014-12-05 18:53 GMT+02:00 Gareth Blades <[email protected]>: > On 05/12/14 16:46, Olli Heiskanen wrote: > > INVITE that Asterisk (at port 5070) receives: > PU.BL.IC.IP:5060 > PU.BL.IC.IP:5070: SIP, length: 1046 > INVITE sip:[email protected];transport=UDP SIP/2.0 > Record-Route: <sip:PU.BL.IC.IP;lr=on;ftag=41030177> > Via: SIP/2.0/UDP > PU.BL.IC.IP;branch=z9hG4bKd7b.ca8b6ac6a82d605cf658af0fea7c9e86.0 > Via: SIP/2.0/UDP > AST.ER.ISK.IP:38699;rport=38699;branch=z9hG4bK-d8754z-bd00e9fd46368417-1---d8754z- > Max-Forwards: 69 > Contact: <sip:[email protected]:38699;transport=UDP> > <sip:[email protected]:38699;transport=UDP> > To: <sip:[email protected];transport=UDP> > From: "771"<sip:[email protected];transport=UDP>;tag=41030177 > Call-ID: YWYwMjMwMmZlODEwM2MwODdjZWJmYjc2NjM5ZmIyNzk. > CSeq: 2 INVITE > Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, > SUBSCRIBE > Content-Type: application/sdp > Supported: replaces, norefersub, extended-refer, timer, > X-cisco-serviceuri > User-Agent: Z 3.2.21357 r21367 > Allow-Events: presence, kpml > Content-Length: 239 > > v=0 > o=Z 0 0 IN IP4 AST.ER.ISK.IP > s=Z > c=IN IP4 AST.ER.ISK.IP > t=0 0 > m=audio 8000 RTP/AVP 3 110 8 0 98 101 > a=rtpmap:110 speex/8000 > a=rtpmap:98 iLBC/8000 > a=fmtp:98 mode=20 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-15 > a=sendrecv > > > This client is saying it only supports speex and iLBC and would prefer > them in that order. > Your sip.conf appears to only permit alaw, ulaw and gsm so there is no > mutual supported codec and hence the call fails. > > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >
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