Comments (3)
Have not tested that scenario, so can't really help a lot. After the iceRestart the WebRTC client sends a SIP reINVITE? Is this reINVITE responded to by the other side (PSTN)?
from webrtc-to-sip.
@havfo yes after ice restart i send the sdp to the other side. the other side does respond to it with a 200OK, but then audio stops flowing. I do observe that rtpengine detects dtmf on the webrtc leg.
Maybe my PSTN gateway provider does not support reinvite
Is it possible to reinvite without relaying it to the other side? it seems like the reinvite does provide the fix for the jumping network on webrtc side but breaks the audio on the sip gateway side.
from webrtc-to-sip.
Actually rtpengine seems to detect dtmf from both side after the reinvite but it just doesnt comeout of either end.
from webrtc-to-sip.
Related Issues (20)
- Build doesn't work HOT 2
- SIP Client with WebRTC SDP HOT 1
- Parallel call forking case HOT 6
- Error!! Unmet build dependencies : libbcg729-dev
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- Add dispatcher HOT 4
- When lets encrypt permission denied
- No audio from SIP to WS HOT 2
- Integration with Asterisk HOT 1
- NO_PUBKEY Error in debian 10 HOT 1
- Testtest
- could not login sipml5 demo to kamailio via websocket. HOT 3
- Sip WIse installation in ubuntu 18.04 HOT 1
- Kamailio + RTPEngine + TURN server behind NAT HOT 5
- webrtc solution for sip protocol
- SIP to WebRTC calls only one way audio HOT 5
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- E: Package 'ngcp-keyring' has no installation candidate
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from webrtc-to-sip.