FYI, I think what you might be seeing in sip.js demo is that webrtc servers try to get clients to bridge RTP directly if possible. We have Asterisk configured to not do this, so that RTP to Halef always goes thru Asterisk. But onsip server likely tells clients to talk RTP directly. I tried sipml5 clients via onsip server and audio works and video connects, but only local video is shown.
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Hi,
FYI, I think what you might be seeing in sip.js demo is that webrtc servers try to get clients to bridge RTP directly if possible. We have Asterisk configured to not do this, so that RTP to Halef always goes thru Asterisk. But onsip server likely tells clients to talk RTP directly. I tried sipml5 clients via onsip server and audio works and video connects, but only local video is shown.