[asterisk-dev] RTCP feedback for codec modules

Matt Fredrickson creslin at digium.com
Tue Oct 10 16:30:15 CDT 2017


On Mon, Oct 9, 2017 at 9:01 AM, marek cervenka <cervajs2 at gmail.com> wrote:

> hi,
>
> i'm writing article about new features in Asterisk 15
>
> can you explain if
>
> https://issues.asterisk.org/jira/browse/ASTERISK-26584
>
> is only part of the building block for function "change codec or codec
> params if network is worse/better"?
>
> and the missing part is rtp_engine + sip update method ?
>

Hey Marek,

This is support added to Asterisk to allow RTCP feedback messages to
influence codec behavior - so statistics contained within those messages
might include packet loss, jitter information, and other network level
details.  The codecs then could presumably act on that behavior (increasing
FEC level, reducing bitrate, etc).  This is separate from any SIP protocol
level messaging.

-- 
Matthew Fredrickson
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-dev/attachments/20171010/fa2e8c29/attachment.html>


More information about the asterisk-dev mailing list