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 -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.pjsip.org/pipermail/pjsip_lists.pjsip.org/attachments/20100406/c5ceac37/attachment.html>
