[Spce-user] NGCP as Web2SIP gateway
alexey.rybalko at gmail.com
Thu Jul 11 07:25:21 EDT 2013
Present implementation of WebRTC is a headache if someone making the bridge
to a legacy SIP cloud. SRTP, SDP media-profiles...you know. Since the
version 2.3.0 of your ngcp-medaproxy-ng Sipwise feeds our hope to get a
relief completing that mission of service convergence :) Playing a couple
of days with SPCE with no valuable success. More questions appear.
The setup is:
1) sip:provider CE,
2) JsSIP as ref. implementation of SIP/WS-client
3) OverSIP as SIP proxy for websocket clients (SPCE comes without
websocket.so, but some reconfiguration is required either if this module
4) softphone (like X-Lite or Jitsi)
The first problem was that Kamailio registrar can't save "Path" header from
JsSIP REGISTER request. Extending DB field "kamilio.location.path" from 128
to 255 characters made that work :) Not sure this is a proper trick but it
works: JsSIP cant register and send INVITE to a "legacy" softphone. But
that's all. The problems are:
1) SPCE (Kamailio) can't locate JsSIP to forward INVITE from softphone.
"Path" header support is enabled however. TCPdump shows us round-robin
traffic from 127.0.0.1 to 127.0.0.1. Softphone receives 408 (Request
timeout) from registrar. Should we use websocket.so instead of external
"websocket-able" SIP-proxy? What might be wrong here?
2) JsSIP drops outbound call (to softphone) on negotiation phase with 488
(Not Acceptable Here). No doubt that's because of unsupported SDES-SRTP and
RTP/SAVPF on remote endpoint. ngcp-mediaproxy is intended to solve this
*"..the complete SDP body is passed to mediaproxy-ng, rewritten and passed
back to Kamailio. Several additional features are available with this
protocol, such as ICE handling, SRTP bridging, etc."
Sounds good. However help is needed to move further. Any input would be
p.s. Is there some manual about packet processing inside mediaproxy-ng?
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