There has been a lot of interesting work in Bayonne recently for implementing phonesystem-like features and functionality, particularly in the SIP driver. None of it has been documented to date.
This is an interesting area of development, and one where Bayonne has certain unique characteristics. One of them is that we fast proxy rtp call connections, which has much lower latency than transcoding. Since we establish the call session at the time of server connect, this imposes some limitations in SDP negotiation since we may later join the call to another device. I have thought of moving the server connection to the join state and to allow SDP to be proxied directly between the endpoints rather than negotiated by bayonne. This would also allow for pure peer calling (where no voice goes through bayonne), perhaps when on the same network. This is something that may get added in the future. I was thinking of starting a new Faq on the wiki related to specific admin/configuration/uses of Bayonne for phone systems. I think if I do a new FAQ, it should explain and offer example of things like, for example, how to configure and use a service provider, like for example broadvoice, for SIP trunking, or how to trunk over SIP with an Asterisk server. It should also explain how we do anonymous calling to allow one to receive calls through public uri's. It probably should also explain how we handle public sdp's and port-forwarding behind NAT, taking advantage of the fact Bayonne allocates RTP sockets in a block of ports.
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