SDP Negotiation

[Date Prev][Date Next][Thread Prev][Thread Next][Date Index][Thread Index]

 



Hi!

On Thu, Jun 12, 2008 at 4:56 PM, Benny Prijono <bennylp at pjsip.org> wrote:

> On Thu, Jun 12, 2008 at 2:22 PM, Sasa Coh <sasacoh at gmail.com> wrote:
> > Hi,
> >
> > Thanks Benny, I got it. Well, we have no To-tag change and unreliable
> > transport.
> > Is there any other SIP scenario/call flow (supported by pjsip) that would
> > handle such functionality?
> > Scenario again: The client receives (after 18x) tones or announcements
> from
> > media server and when other party answers (2xx) the SDP is changed.
> >
>
> Since the To-tag doesn't change, probably we can try this. Basically
> we will force the server to not change the SDP between 18x and 2xx
> response, by using reliable response. Hopefully this will make the
> server sends the new SDP with UPDATE or re-INVITE, which we support.
>
> You can force the server to send answer reliably either with using TCP
> or TLS transport, or using reliable provisional response extension
> with --use-100rel option (in pjsua).
>
> If the server still sends modified SDP in 2xx, at least we can then
> blame it for being non-compliant. ;-)


Thanks a lot Benny. That's exactly what I need :-)

bye,
sasa


>
> Cheers
>  Benny
>
>
> > thanks again,
> > sasa
> >
> > On Thu, Jun 12, 2008 at 2:52 PM, Benny Prijono <bennylp at pjsip.org>
> wrote:
> >>
> >> On Thu, Jun 12, 2008 at 11:59 AM, Sasa Coh <sasacoh at gmail.com> wrote:
> >> > Hello Benny!
> >> >
> >> > I have question regarding SDP negotiation process. We managed to
> >> > establish
> >> > signaling connection with the other party but there was no (correct)
> >> > voice
> >> > path established.
> >> >
> >> > Here is how it goes:
> >> >  - pjsua makes call through Call Agent: INVITE with SDP (offer).
> >> >  - Call Agent responds with 180 Ringing with SDP (answer). Here SDP
> >> > points
> >> > to the Media Server that provides "Ring Back" tone to the caller.
> >> >  - When called party accepts the call, Call Agent sends 200 OK with
> yet
> >> > another SDP (offer/answer?) points to the called party.
> >> >  - After that there is no voice path? to be exact, voice path is still
> >> > established to the media server providing tones but not to the called
> >> > party
> >> > as directed in the latter SDP.
> >> >
> >> > The same situation works with other clients (X-Lite).
> >> >
> >> > Does pjsip support this kind of SDP negotiation? Should we handle some
> >> > callback we are not aware of?
> >> >
> >>
> >> I think this is quite a complicated scenario, and which one it is
> >> depends on whether the server changes the To tag between 18x and 2xx
> >> response. If it does change, then this is a similar scenario to
> >> forking with early media, which we don't handle at PJSUA-LIB level (we
> >> do have basic forking handler at PJSIP level, but I don't implement
> >> forking at PJSUA-LIB level due to complexities in handling the media).
> >>
> >> If the To-tag doesn't change, then it further depends on whether the
> >> SIP transport is reliable or not. If the transport is not reliable, it
> >> means server is sending "provisional" SDP in 18x response and the
> >> "final" SDP in the 2xx response which the UAC should use. This, albeit
> >> hairy is a valid SIP scenario. But we don't seem to handle this
> >> either. If the transport is reliable, then the SDP in 18x is the final
> >> one and UAC must ignore the SDP in 2xx since it belongs to the same
> >> transaction, and if this is the case, PJSIP is doing the right thing.
> >>
> >> So there you go. :)
> >>
> >> Cheers
> >>  Benny
> >>
> >> _______________________________________________
> >> 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/20080613/d5a4b8d4/attachment.html 


[Index of Archives]     [Asterisk Users]     [Asterisk App Development]     [Linux ARM Kernel]     [Linux ARM]     [Linux Omap]     [Fedora ARM]     [IETF Annouce]     [Security]     [Bugtraq]     [Linux]     [Linux OMAP]     [Linux MIPS]     [Linux API]
  Powered by Linux