[reSIProcate] Asterisk 11 + repro WebRTC tested

[Adam Roach]

Awesome work, Daniel.

Out of curiosity, did you try this with Firefox also? WebRTC is on by 
default in Nightly, Aurora, and Beta now, and can be activated in 
Production builds by going to about:config and setting 
media.peerconnection.enabled to "true." (Basically, it's on by default 
in Firefox 22 and later). It would be great if we saw cross-browser 
compatibility for this kind of setup.

If you run into any problems with Firefox in this configuration, please 
let me know. I'd like to prioritize real-life interop bugs in our WebRTC 
work.

/a


On 6/3/13 16:38, Daniel Pocock wrote:
>
>
> 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:
>
> BackGround(/usr/share/asterisk/sounds/demo-congrats)
>
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