mailsvb wrote:
Hi everyone,
Hola!
I'm currently trying to play a little with WebRTC using sipml5 client and Asterisk trunk (370821) It seems the the WebRTC implementation for Asterisk 11 is already available in the trunk? Am I right? http://lists.digium.com/pipermail/asterisk-dev/2012-July/055940.html
You are correct but presently the media side has been untested fully since Google Chrome does not yet implement ICE according to the RFC. This won't affect your registration attempt, though.
I'm having trouble to even register to my Asterisk server using sipml5 client. The only reference to websockets in the sample config files is in sip.conf, but it seems to be for when Asterisk needs to register somewhere ;register => tls://username:[email protected] <mailto:username%[email protected]> ; ; The'transport' part defaults to'udp' but may also be'tcp','tls',*'ws', or'wss'*. ; Using'udp://' explicitly is also useful in case the username part ; contains a'/' ('user/name').
In the sip.conf entry for the account you are trying to register as place the following:
transport=ws
How do I set up a user in Asterisk so that I can register via sipml5? Attached you can find a wireshark trace from my register attempts, first on TCP port 5060, than on the Asterisk http server default port 8088...
You will need to change sipml5 to use http://<hostname or IP address of Asterisk>:8088/ws as the URL. WebSocket is only available on the /ws path.
-- Joshua Colp Digium, Inc. | Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com & www.asterisk.org -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
