Hi, Use the stram util example.... which needs a tweak for using the multicast support which is there in some pjsip discussions. Using that generate a conference port for thta stream..... and when an incoming call comes in just conference these two streams... Ravi On Tue, Apr 6, 2010 at 1:37 PM, Gao Li Yang <gaoly at ghtchina.com> wrote: > Hi, > > > > I?m going to implement a SIP user agent which can support multicast paging > using PJSIP. This means, when an incoming call arrives, > > this UA should answer it automatically and put all received RTP packages to > the preconfigured multicast address/port, > > that will be a one-way RTP stream and I know some SIP phones like Aastra > 53i can listening on these multicast addresses > > and automatically play the RTP stream on its speaker. > > > > Because I don?t have any experience on PJSIP, I wonder does anyone have > similar development experience, or have any idea about > > how to use PJSIP to implement such a function? I think probably the main > task will be transferring all received RTP packets to a specific multicast > address. > > > > Thanks in advance. > > > > ~Gary > > > > > > _______________________________________________ > 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/20100406/c7a49402/attachment.html>
