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When I browse to my instance I am unable to sign in , I have tried from Firefox on Linux and OSX as well as Chrome on OSX and Android. I only see an error when using Firefox which is:
Janus error: er…
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I have tried the online demo and running the demo on my own box (Azure VM) with Janus and DTMF tones are not working on my test outbound calls. Is there something that needs to be configured for DTMF…
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I have been toying with the idea of creating a thirdparty call control/b2bua/native client behind the gatway. As a prototype, i created a simple python script to register the user using videocall plug…
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I just wanted to offer this example build script, which works on ubuntu 14.04 and debian 8 (and probably later ubuntu versions as well), and builds and uses boringssl, libwebsockets, and latest libsrt…
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It looks like there may be a reference leak in old_sessions in the sip plugin.
In my logs I see:
Checking 1 old SIP sessions...
Checking 1 old SIP sessions...
Cleaning up handle 2390786515...
[23907…
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Hello,
I've been running the janus-gateway for 28 hours yesterday and ran into the following error:
[31m[ERR][0m [dtls.c:janus_dtls_srtp_incoming_msg:311:] Handshake error: error:1410241B:SSL routin…
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when i register a sip client i get this error
```
nta: bind(XXX.XXX.XXX.XXX:*;transport=*): Protocol not supported
nua: initializing SIP stack failed
```
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A version of janus from June 11 (11a33f5cad1d71ca396c6f8da02f813d9654b0f8) works with both firefox and chrome both publishing and viewing.
Current master seems to work with firefox publishing, firefo…
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I've been debugging a weird issue which in a nutshell consists of ICE failing for outgoing calls, even when it shouldn't.
After a lot of digging I narrowed it down to the fact that ICE is started too…
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I installed Janus-WebRTC gateway on ubuntu server and started created a multiuser chat and broadcast. When is run this example Janus echotest example In my own server i always getting a bad resolution…