Browser to Browser call in Asterisk

Hi Everyone,

From the below link i could achieve audio prompt files in SIPML5(Webrtc - Asterisk)

And my question is that whether i can achieve audio call(browser to browser call) with Asterisk and if so please provide some links to achieve it.

There is nothing different between calling from a WebRTC client to a WebRTC client and calling from a desk phone to a desk phone. They both work the same way within Asterisk, so any documentation for that is applicable. You really just need to use Dial() in the dialplan to call the other client.

Thanks jcolp for your response

But can you say how I should configure in extensions.conf. I tried with below extensions.config but it didnt connect the call.

exten => _X.,1,Dial(SIP/${EXTEN})
exten => _X.,n,Hangup()

getting below errors
May 10 18:29:54] ERROR[32099][C-00000007]: netsock2.c:305 ast_sockaddr_resolve: getaddrinfo(“df7jal23ls0d.invalid”, “(null)”, …): Temporary failure in name resolution
[May 10 18:29:54] WARNING[32099][C-00000007]: chan_sip.c:16762 __set_address_from_contact: Invalid host name in Contact: (can’t resolve in DNS) : ‘df7jal23ls0d.invalid’
[May 10 18:29:54] ERROR[32099][C-00000007]: netsock2.c:98 ast_sockaddr_stringify_fmt: getnameinfo(): ai_family not supported
Audio is at 10960

What version of Asterisk are you using, and is the client still connected to Asterisk at the time?

I am using Asterisk version 14.4.0 and Client is SIPML5

I haven’t used chan_sip for WebRTC in years so I can’t comment on specifics (someone else may be able to) but is the websocket connection from the browser to Asterisk still up when you attempt to call? That’s the only thing that comes to mind, otherwise there may be something else with your configuration (I haven’t seen any issue reports in recent times or anyone mention this).