Joel Dodson wrote: > Hi Robert, > > I used PJSIP/PJMEDIA to do what sounds like almost exactly what you're > wanting to do. I used the lower layer APIs, not pjsua. > > I also used the rtp module in PJMEDIA to directly access the rtp and > rtcp packets. I didn't use the higher level audio port concept. > > Check out siprtp.c and simpleua.c in pjsip-apps/src/samples. Those > are good examples of how to use the lower level APIs and especially > the pjsip_inv_session and pjmedia_rtp_session (disclaimer, I'm looking > at pjsip 1.6 release, I haven't used pjsip in a while so I hope those > examples and abstractions are still there). Assuming those > abstractions still exist, they're a great place to start. > > I was able to support several hundred simultaneous g711 based audio > streams through my signaling and media gateway with very low latency. > That was on windows XP running on VMs. You should be able to get much > more than that on a better hardware configuration. > > good luck, > > Joel > > > On Mon, Nov 14, 2011 at 5:40 PM, Robert Reif<reif at earthlink.net> wrote: > >> I need to develop a SIP gateway on Windows to another networked protocol for >> a dynamic number of channels and I am looking at pjsip for the SIP side. >> >> Can I have multiple active users in independent calls and add and subtract >> users at runtime? >> >> Can I get access to the PCM audio streams for each user? >> >> Are there any gateway examples available? >> >> I just found this project today and have been reading the documentation and >> source code. I built the code and am using it for interoperability testing >> and it looks like it will do what I need but I would like some conformation >> or advice before I invest a lot of effort. >> >> _______________________________________________ >> 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 > > Thanks Joel, It's good to know that pjsip can do what I need. However I would like to work at a higher level because I don't want to restrict what codecs are supported on the SIP side. I would also like to take advantage of the jitter buffer so I don't want to work with the RTP packets. I may also have the need in the future to support multiple users calling into a single channel so I might need access to the combiner. The audio format on the other side of the gateway can be any format, packet size and sample rate and will probably be different than the SIP side. The library for the other side dynamically handles the buffering, jitter, SRC, format conversions and combining for it's protocols so all it needs is the raw PCM audio stream per channel.
