I’ve just done a test with a WebRTC client connecting to the repro proxy with the SIP messages relayed over TCP to Asterisk
Asterisk successfully answers the call using SAVPF, SRTP and ICE.
The client is greeted by the demo
This was tested in the Asterisk 11 environment described in my earlier email about SRTP build issues on the asterisk-users list.
This is quite useful because it proves that Asterisk doesn’t have to be exposed as the HTTP WebSocket server: all the WebSocket handshake and message parsing is done by the proxy.
Specific versions tested:
– Asterisk 11.4 built from SRPM on CentOS 6 + EPEL6
– repro 1.9.0~alpha0 package from Debian experimental
– JsSIP `tryit’ client
– Google Chrome
Just some more notes about problems encountered with the Asterisk SRPM:
it doesn’t seem to know anything about /usr/share/asterisk/sounds – even though I install both the gsm and ulaw sounds RPMs, it always gives errors such as file.c:701 ast_openstream_full: File demo-congrats does not exist in any format
I manually edited extensions.conf to include the full absolute paths and then it works, e.g: