[Spce-user] Webrtc to SIP JSSIP
rich at e-isystems.com
Sun May 4 02:00:46 EDT 2014
I've been trying to get Webrtc to work using the JSSIP package. Using the http://tryit.jssip.net/ demo, I can get 2 Chrome browser sessions to connect with audio and video. However, when I disable video and try to connect to a SIP softphone such as Jitsi I'm having no luck. I'm using the spce mr3.2.1. Here's what I've tried so far.
I've changed the "NAT and Media Flow" parameters for both extensions using a bunch of combinations without success. I've tried using Jitsi with UDP, with TLS, with opus enabled, SAVPF enabled. I got the Jitsi to answer the call, but no audio will flow. I've also tried connecting to Bria on my iPhone and it doesn't even try. Just fails with 488 error. Can anyone help explain how the mediaproxy-ng is involved here? Maybe I need some settings changed in spce. Does Mediaproxy-ng have the ability to transcode between SAVPF and RTP?
It would be great to do all combinations of Webrtc to SIP and back. But I would settle for Webrtc to SIP endpoints only. My requirements really have no need for Webrtc to Webrtc. Which of course is the only thing I got working. ;)
I appreciate any help.
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