Hi Anshuman, I did something very similar a while ago. I wrote a signaling and media gateway to go from a proprietary signaling and RTP-like network to a standard SIP/RTP network. Both sides were using g711[u|a]law though. The big problem was the packetization period was different between the two sides. No audio needed to be decoded, just repacketized. I didn't use the conferencing facilities, I went directly with the RTP support in PJMEDIA. For an example, look at: pjsip-apps/src/samples/siprtp.c That's one of the samples I used as a reference for how I handled RTP. I don't recall the exact numbers, but we had very low latency even though we did have to do some buffering to accommodate the difference in pacetization period. good luck. Joel On Wed, Apr 27, 2011 at 9:16 AM, Anshuman S. Rawat <arawat at 3clogic.com> wrote: > Hi, > > Has anyone attempted to use PJSIP as a media-relay? Use case is - a VOIP > application using pjsip dials out 2 numbers and relays media between the 2 > parties. I am aware that the conferencing facility can be used to achieve > this but that will involve encoding/decoding media from both parties and I > want to avoid that and make PJSIP work purely as a relay. > > What are the difficulties seen with this? > > Thanks, > Anshuman > _______________________________________________ > 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 > >
