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sip - No audio in WebRTC and Asterisk - Stack Overflow
https://stackoverflow.com/questions/28501309/no-audio-in-webrtc-and-asterisk#:~:text=Starting%20with%20Asterisk%2012%20you%20need%20to%20have,in%20your%20WebRTC%20calls%20and%20no%20warning%20whatsoever%21
Asterisk webrtc no audio - Asterisk WebRTC - Asterisk ...
https://community.asterisk.org/t/asterisk-webrtc-no-audio/91333
I config asterisk for webrtc and sipjs.com. so that work for SimpleUser configuration. with SimpleUser every thing is Ok. but when I use UserAgent , connection has established without audio .this is asterisk console when User3 call User2 : WebSocket connection from '2.185.145.48:45486' for protocol 'sip' accepted using version '13' -- Added contact …
sip - No audio in WebRTC and Asterisk - Stack Overflow
https://stackoverflow.com/questions/28501309/no-audio-in-webrtc-and-asterisk
No audio in WebRTC and Asterisk. Ask Question Asked 6 years, 11 months ago. Active 1 year, 6 months ago. Viewed 6k times 0 I have a strange issue with Asterisk (in this case 13.2 version) and WebRTC. So, I have latest Asterisk 13.2, latest Crome (with Firefox - same problem) and sip.js (also tried with sipml5) and local network - no nat or ...
Asterisk 16.4 & WebRTC = no audio - Asterisk WebRTC ...
https://community.asterisk.org/t/asterisk-16-4-webrtc-no-audio/83624
Hello there, we’ve just tried our fist webrtc test with Asterisk 16.4.0 The endpoints get registered well and it’s able to send calls without problems … but without audio. This is the Asterisk CLI: -- Executing […
WEBRTC no audio - Asterisk WebRTC - Asterisk Community
https://community.asterisk.org/t/webrtc-no-audio/90896
WEBRTC no audio. Asterisk Asterisk WebRTC. bobtarask December 11, 2021, 6:19pm #1. Hello there ! I’m trying to use WEBRTC with SIPJS and Asterisk (Asterisk GIT-16-89cf7899be) . It works very well with Chrome version ~50. Since I’m using chrome 96 on Android 10 audio seems not working at all…. Maybe codecs change?
Asterisk: WebRTC no audio - Asterisk WebRTC - Asterisk ...
https://community.asterisk.org/t/asterisk-webrtc-no-audio/82009
Calls between two SIP clients (zoiper) are successful. When I start a call between a WebRTC client (sipml5) and a SIP client (Zoiper) is the connection active, but there is no audio in both directions available. So, I have latest Asterisk 16, latest Chrome (with Firefox & Chrome Beta the same problem), sipml5 and a local network - no nat or ...
web - WebRTC on standalone asterisk - no audio - Server Fault
https://serverfault.com/questions/671241/webrtc-on-standalone-asterisk-no-audio
Im not getting audio from WebRTC to WebRTC clients. I work in a LAN environment. Situation - Call from JSSIP to JSSIP (same client) with a standalone Asterisk server. Problem. There is no audio at all when doing a call from 6001(JSSIP) to 6002(JSSIP). After a while some RTP packets are getting send, but not received.
No audio on PJSIP channels - Asterisk WebRTC - Asterisk ...
https://community.asterisk.org/t/no-audio-on-pjsip-channels/90281
Hello, I don’t understand why I have no audio on my endpoints, When I log my softphones from a local address, everything works fine, but if I log from outside my network I have no audio. I don’t understand how Astertisk can match the local IP address of my client, since In my softphone code I use my public IP address to log to asterisk I’m using WebRTC for my …
Asterisk and JsSIP no audio - Asterisk WebRTC - Asterisk ...
https://community.asterisk.org/t/asterisk-and-jssip-no-audio/84312
Hi, I’m connecting a webrtc client to Asterisk 16, but I can’t hear the audio playback (dialed 200, and I should hear the demo-congrats audio file). Connecting with a SIP softphone works fine. The connections are within the same LAN segment. This is my pjsip.conf: [transport-wss] type=transport protocol=wss bind=0.0.0.0 ; All other transport parameters are …
Webrtc SipML 5 audio issue + asterisk - Asterisk WebRTC ...
https://community.asterisk.org/t/webrtc-sipml-5-audio-issue-asterisk/83737
Hi, We are using asterisk 16.0.7 + webrtc for video calling but facing the issue of no audio in case when caller dial to callee… in this case we ain’t get any audio at callee side. if caller (audio YES/ video YES)-------> callee.(video YES , audio NO) can any one suggest were we are doing wrong below is our configuration : pjsip.conf [webrtc_client] type=aor max_contacts=1 …
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