On 05/02/14 19:45, Kurt Roeckx wrote: > On Wed, Feb 05, 2014 at 07:29:24PM +0100, Daniel Pocock wrote: >> >> >> On 04/02/14 23:43, Kurt Roeckx wrote: >>> On Tue, Feb 04, 2014 at 10:15:42PM +0100, Daniel Pocock wrote: >>>> >>>> >>>> I'm just wondering if people have further feedback about the Debian SIP >>>> service >>>> >>>> https://rtc.debian.org has just been updated to overcome some issues >>>> with local audio feedback >>>> >>>> For Firefox/Iceweasel users, there have been fixes in both the >>>> JSCommunicator code and the Asterisk instance behind >>>> http://www.sip5060.net/test-calls so this should also be stable. >>>> >>>> If there are complaints that haven't been answered, please let me know. >>> >>> I actually can't get those to work. That is, I can call them, >>> they pick up, but I never hear sound from the other side. >>> >> >> Please open the JavaScript console before starting the call to the test >> number > > This was with jitsi >
OK, this may be because Jitsi doesn't currently use ICE/TURN with SIP It does use ICE/TURN with XMPP (all the protocol support is there behind the scenes) and they are planning to extend it to SIP very soon now: https://jitsi.org/Documentation/FAQ#stun In fact I'm very confident they can make this work because I borrowed their ice4j.jar and put it in Lumicall, where I have it working with SIP for 2 years now. -- To UNSUBSCRIBE, email to [email protected] with a subject of "unsubscribe". Trouble? Contact [email protected] Archive: http://lists.debian.org/[email protected]
