I think the issue is that the underlying pjsip libs are behaving in a standards compliant way (SDP with just telephone events appears valid to me), but the pjsua app can't complete the call because it expects media. Bill On 12/3/2014 4:14 AM, Harald Radke wrote: > Hey Bill, > well I dont have a problem with sessions depending on a codec > definition to be accepted, I was irritated that pjsip seems to treat > the "telepone-event" codec differently compared to "normal" ones when > it comes to mismatch detection. > a) If UA1 has e.g. a G711 codec and gets an INVITE with a L16 codec > (with or without an additional telephone-event codec) definition in > the media, pjsip_inv_answer() for a 200 response will return with > PJMEDIA_SDPNEG_NOANSCODEC, which is fine. > b) But if UA1 gets an INVITE with the telephone-event codec only > defintion, the SDP negotiation returns successfully, produces that > (IMHO strange) SDP media entry with only type, port and transport type > but no codec defintion and leaves the inviting party to figure that > the INVITE actually failed. > I would have expected a consitent behaviour for a) and b), with the > SDP negotiation failing on the listener side > Regards, > Harry > *Gesendet:* Dienstag, 02. Dezember 2014 um 18:53 Uhr > *Von:* "Bill Gardner" <billg at wavearts.com> > *An:* pjsip at lists.pjsip.org > *Betreff:* Re: [pjsip] telephne-event > Hi Harry, > > The issue is lack of audio. The caller has no codec defined. This is > OK from SIP perspective, but pjsua library depends on having audio in > order to start a call session, at least it did a few years ago when I > looked into this. The only way to start a call to pjsua without audio > is for the invite SDP to specify a codec and also include a=inactive > line, which says the audio is inactive. > > Bill > > On 12/2/2014 10:01 AM, Harald Radke wrote: > > Hi there, > I just stumbled over some situation: > - two slightly modified instances of > pjsip-apps\samples\src\simpleua.c: > UA1: > SIP Port 15060 > RTP Port 14000 > G711 codec, telephone-events DISABLED > UA2: > SIP Port 25060 > RTP Port 24000 > NO codec, telephone-events ENABLED > UA2 connects to UA1: > ------------------------------ > INVITE sip:127.0.0.1:15060 SIP/2.0 > [...] > Content-Type: application/sdp > Content-Length: 215 > v=0 > o=- 3626519662 3626519662 IN IP4 127.0.0.1 > s=pjmedia > t=0 0 > m=audio 24000 RTP/AVP 96 > c=IN IP4 127.0.0.1 > a=rtcp:24001 IN IP4 127.0.0.1 > a=sendrecv > a=rtpmap:96 telephone-event/8000 > a=fmtp:96 0-16 > UA1 fails to set up stream: > ----------------------------------- > Unable to create audio stream info: Invalid media payload type > (PJMEDIA_EINVALIDPT) [code=220102] > but still sends OK response, however kinda incomplete: > ----------------------------------------------------------------------- > SIP/2.0 200 OK > [...] > Content-Type: application/sdp > Content-Length: 176 > v=0 > o=- 3626519662 3626519663 IN IP4 127.0.0.1 > s=pjmedia > t=0 0 > m=audio 14000 RTP/AVP > c=IN IP4 127.0.0.1 > b=TIAS:64000 > a=rtcp:14001 IN IP4 127.0.0.1 > a=sendrecv > UA2 sends a BYE since the SDP data is broken: > ------------------------------------------------------------ > inv0x246caf8 ....SDP offer/answer incomplete, ending the session > endpoint .....Request msg BYE/cseq=18248 (tdta0x2478200) created. > so the session is terminated, however I would have expected it to > be terminated by the listener, (pjsip_inv_answer() for the OK > response not succeeding, just as it is in case of non-matching codecs) > Any thoughts? > Regards, > Harry > > _______________________________________________ > Visit our blog:http://blog.pjsip.org > > pjsip mailing list > pjsip at lists.pjsip.org > http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org > > > _______________________________________________ Visit our blog: > http://blog.pjsip.org pjsip mailing list pjsip at lists.pjsip.org > http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org > > > _______________________________________________ > Visit our blog: http://blog.pjsip.org > > pjsip mailing list > pjsip at lists.pjsip.org > http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.pjsip.org/pipermail/pjsip_lists.pjsip.org/attachments/20141203/27e38224/attachment.html>
