Joel, First of all thanks for such a detailed email and for educating me as well.Yes, B2BUA is exactly what I am looking for. I will do my ground work with the leads and pointers that you have provided to me to see where I go from here. Cheers, Senthil. ----- Original Message ---- From: Joel Dodson <jdodson@xxxxxxx> To: pjsip list <pjsip at lists.pjsip.org> Sent: Friday, April 4, 2008 10:44:01 AM Subject: Re: Need help on getting rtp stream on sip server Hi Senthil, I'm currently working on a gateway that's doing half of what you're asking about. It sounds like your server will need to be a B2BUA to terminate the media on either side and connect the two streams. I'm using the invite session abstraction to perform the offer/answer mechanism to exchange SDPs to get the endpoint to send the media to my gateway. Invite session is a great abstraction. It's documented in the PJSIP developers guide. For an example, look at simpleua.c. For how to get the media out of the stream and send it to the other stream, look at the pjmedia documentation for the pjmedia_port interface. The stream interface is great if you'll potentially have different codecs on either side of the call. If you'll have the same media parameters on each side, you probably just want to forward the RTP. If you want to get the RTP directly, look at siprtp.c. siprtp.c isn't using the jitter buffers, it's mainly capturing the RTP and looking at stats. There's a very simple interface to the jitter buffers though which is also well documented, so it shouldn't be too hard to plug that in to your code. If you're simply forwarding RTP though, you shouldn't need the jitter buffer. Hope the helps. Joel On Thu, Apr 3, 2008 at 11:52 PM, Senthil Raja <vsraja at yahoo.com> wrote: > > Hi Benny, et al, > With your help (thanks a lot!) and from the sample stateful_proxy.c, I was > able to create sip server which does registrar and proxy functions. This > setup allows two soft-phones to registrar themselves and make calls between > them. However the rtp stream for the call is established directly between > the User Agents and not via the sip server. Hence I would like to know > what changes should I have to perform to make the rtp go through the sip > server. Appreciate your help. > > Thanks, > Senthil. > > > ________________________________ > You rock. That's why Blockbuster's offering you one month of Blockbuster > Total Access, No Cost. > _______________________________________________ > 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 ____________________________________________________________________________________ You rock. That's why Blockbuster's offering you one month of Blockbuster Total Access, No Cost. http://tc.deals.yahoo.com/tc/blockbuster/text5.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.pjsip.org/pipermail/pjsip_lists.pjsip.org/attachments/20080404/a531147c/attachment.html
