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One way audio with WebRTC and Asterisk 15 - Stack …
https://stackoverflow.com/questions/54201693/one-way-audio-with-webrtc-and-asterisk-15
Show activity on this post. I have two ways audio when calling from a WebRTC client connected to Asterisk to my mobile phone, but when I call from the mobile phone to the WebRTC client the call is established and there is only one way audio: from the WebRTC client to the mobile phone. I'm using Asterisk 15.6.1 installed on a VPS with static IP, the WebRTC client …
One way audio with Asterisk 15 and a WebRTC client ...
https://community.asterisk.org/t/one-way-audio-with-asterisk-15-and-a-webrtc-client/77955
The result is the same: two ways audio when calling from the browser to the PSTN and one way audio when calling from the PSTN to the browser. My debug lines look like: <— Received SIP request (428 bytes) from WSS:152.231.162.25:53283 —> BYE sip:asterisk @ mail . example . com:5060;transport=ws SIP/2.0
One way audio in webrtc configured asterisk server ...
https://community.asterisk.org/t/one-way-audio-in-webrtc-configured-asterisk-server/82327
As well please enable SIP tracing in the log and check the negotiated audio ports (a pcap would suffice as well). Then enable rtp debug in Asterisk: *CLI> rtp set debug on And compare the ports for incoming, and outgoing audio to make sure audio is flowing to/from the expected ports using the negotiated codecs.
[asterisk-users] One Way Audio with WebRTC (with external ...
https://asterisk-users.digium.narkive.com/wsRleAgi/one-way-audio-with-webrtc-with-external-asterisk
Dealing with just the WebRTC asterisk server, asteriskrtc.local, I am able to log in to the SIPml webpage and make a call from a SIP Phone to that WebRTC user, and vice versa, and all the media flows. - asteriskgary.local user, 1000, dials asteriskrtc.local number, 6901. - 6901 sees the call and has the option to answer. - 6901 answers the call.
[Help] WebRTC one way audio/video problem on Asterisk 12 ...
https://community.asterisk.org/t/help-webrtc-one-way-audio-video-problem-on-asterisk-12/45335
Hello I’m working on WebRTC project with Asterisk 12.0.0 and JsSip 0.3.7 WebRTC client. For my testing purposes I set up a simple testing environment, which contains: Asterisk 12.0.0 (10.10.1.10), rfc5766-turn-server used as STUN server (10.10.1.90) and 2 PC’s used as WebRTC clients (10.10.1.11 and 10.10.1.12). All testing devices are of course behind NAT which is …
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