asterisk-opus
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Opus (transcoding) and VP8 (passthrough) support for Asterisk, needed for a better WebRTC integration
Hi, could you please explain to me the business value of this feature? Asterisk alone is able to use the WebRTC project. So why I should use both Janus and...
I am getting a lot of this: `WARNING[1227][C-0000000c]: chan_sip.c:7184 sip_write: Asked to transmit frame type opus, while native formats is (ulaw) read/write = opus/opus` I am able to see the...
Hi, I noticed that this patch does not compensate for lost packets. May I know the reason why? according to opus API, you should use opus_decode with null pointer when...
Recently compiled Asterisk 11.17 with this patch. The codec itself works great, no issues so far. However, when MixMonitor starts recording my outgoing calls when I'm using Opus codec the...
Anyone with knowledge about this patch could possibly add support to Asterisk 11.6-cert2 version? Thanks :)
Post by Matthew Jordan at asterisk-dev (http://lists.digium.com/pipermail/asterisk-dev/2013-May/060419.html) Hello! I'm going to comment here specifically to clarify Digium's position on Opus and VP8 as codecs and their inclusion in Asterisk. To...
Hey there! I went ahead and updated your patch for the latest release. Feel free to copy/modify/blah/whatever. No need to mention me anywhere, just trying to make it easier for...
I've found that If I compile Asterisk 11.6 or more (I'm actually using Asterisk 11.6-cert1) then there is a conflict with ICE and RTP communication More information here: http://forums.digium.com/viewtopic.php?f=1&t=89389 Maybe...
Hi, I tried the patch, it's working. When I use SipML5 to make a conference call to asterisk, some time video is dirty for participants. There are also a lot...