2009/3/25 Fran?ois Roseberry <frank_pharaoh05 at hotmail.com> > Hi, > > we've found out that our server changed the whole SDP before giving it to > the callee (No matter how we changed it, it was always the same). > The SIP server we used is 3CX Phone System, from Microsoft, which I guessed > is made to work with 3CX softphone, so the replaced SDP does the work for > this. We found another free and more "transparent" SIP server, which ias a > lot easier, more user-friendly, it's called minisipserver. > > That explains it! > Now the SDP passes, and the two media lines are established, and we can > talk and hear each other through the audio line, just as normal. > But we added a dummy video codec (which doesn't exist), rtpmap:2 > BITMAP/16000, before the telephone-events line. Otherwise, it crashed at the > media session creation. > Is that in pjmedia_session_create()? Is it crash or assertion? But if you say that telephony-events attribute is not mandatory, we'll try > to remove both. > > I don't think telephony-events is right for m=video (it doesn't sound like making sense). Sorry I think I was wrong about empty format list in SDP m= line. Syntactically, the spec does allow empty format list, but in pjsip we only accept that if port number is zero. So you need to codec indeed. > We understand what the attributes connection, rtcp and sendrecv (media > direction) are for, but what about fmtp and rtpmap ? > rtpmap is mandatory for dynamic PT. fmtp is additional info about the codec, and it's codec specific. Its use should be described in the corresponding audio/video profile RFC I think. -benny We have a vague idea, since the rtpmap attribute of the already created > audio "m" line contains info about a codec, like PCMU or PCMA, and a > supported bitrate. There's also the telephony-events that we're not sure > about. > > If you could just give us a little hint on that, > > and we'll keep up, > thanks > > Fran?ois Roseberry* > * > > > > > ------------------------------ > Date: Wed, 25 Mar 2009 15:40:24 +0000 > From: bennylp@xxxxxxxxx > To: pjsip at lists.pjsip.org > Subject: Re: SDP Question > > > 2009/3/25 Fran?ois Roseberry <frank_pharaoh05 at hotmail.com> > > Thanks for the quick reply, > > Here's the Wireshark capture. As you can see, we added a connection line in > the second media, even though it's optional. > > > That should be okay, it will be picked up if it's present. > > > And we don't have any video codec. I want to know if we're obliged to have > at least one ? > > > That's fine too. You can also omit the telephone-event and just put empty > format list in the m=video line, which is valid according to the spec. > > At glance, the SDP looks okay, I couldn't find anything wrong with it > either. Perhaps the best way to troubleshoot this is to put breakpoint on > pjmedia_sdp_parse() function in pjmedia/sdp.c, and trace it step by step to > know what the parser thinks about it. I'll do that as soon as time permits. > > cheers > Benny > > > > > > Fran?ois Roseberry* > * > > > > ------------------------------ > Messenger vous offre des tonnes de nouvelles fonctions qui rendent le > clavardage encore plus plaisant Cliquez ici pour en savoir plus.<http://go.microsoft.com/?linkid=9650738> > > _______________________________________________ > 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 > > > > ------------------------------ > Communiquez, mettez ? jour et planifiez sur Windows Live Messenger. D?butez > aujourd'hui. <http://go.microsoft.com/?linkid=9650737> > > _______________________________________________ > 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/20090326/1d2b4b0a/attachment.html>
