[Freeswitch-users] absolute_codec_string not working

[Ken Rice]

If you are limiting the calls to specific codecs and avoiding transcoding, proxy media doesn’t really reduce the overhead anymore… that changed a few years ago but the notion its better still hangs on today

 

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of L?i Ð?ng
Sent: Tuesday, November 22, 2016 9:07 PM
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] absolute_codec_string not working

 

Hi @Michael, you were right, I'm intentionally using media_proxy for FS, since I want to reduce CPU usage on FS machine.

In this case, I just want to limit the codecs used for each endpoint, and codec negotiation will be handled by them.

e.g: caller use PCMA, PCMU, GSM by its own in INVITE, I want to limit the callee to only use PCMA,GSM.

Look like `absolute_codec_string` is not what I'm looking for right? Any way out?




Loi Dang Thanh

Phone : 01224.735.448

Email : loi.dangthanh at gmail.com

 

On Tue, Nov 22, 2016 at 9:57 PM, Michael Jerris <mike at jerris.com <mailto:mike at jerris.com> > wrote:

using proxy_media is my best guess but can’t tell with this little info.

 

On Nov 22, 2016, at 5:27 AM, Lợi Đặng <loi.dangthanh at gmail.com <mailto:loi.dangthanh at gmail.com> > wrote:

 

 

Hi List, I got some trouble with using `absolute_codec_string` param.

My call scenario is pretty simple: caller <--> FS <--> callee.

My caller compose `m=audio 7078 RTP/AVP 8 0 101` in its INVITE, and I'm doing `<action application="bridge" data="{absolute_codec_string=PCMU,GSM}sofia/gateway/callee/$1"/>` in the dialplan.

But FS still use `m=audio 22952 RTP/AVP 8 0 101` in the INVITE to the callee.

not sure what I'm missing, helps would be appreciated.

Note that when I'm using `originate` application in fs_cli, things are good.

`originate {absolute_codec_string=PCMU}sofia/gateway/caller/100 &bridge({absolute_codec_string=PCMA}sofia/gateway/callee/100`. 

I have FS with proper behavior in transcoding, caller has `m=audio 31184 RTP/AVP 0 101` received, and callee has `m=audio 21922 RTP/AVP 8 101` received.

rgds,


Loi Dang Thanh

Phone : 84.1224.735.448

Email : loi.dangthanh at gmail.com <mailto:loi.dangthanh at gmail.com> 

_________________________________________________________________________
Professional FreeSWITCH Consulting Services: 
consulting at freeswitch.org <mailto:consulting at freeswitch.org> 
http://www.freeswitchsolutions.com

Official FreeSWITCH Sites
http://www.freeswitch.org
http://confluence.freeswitch.org
http://www.cluecon.com

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org> 
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

 


_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org <mailto:consulting at freeswitch.org> 
http://www.freeswitchsolutions.com

Official FreeSWITCH Sites
http://www.freeswitch.org
http://confluence.freeswitch.org
http://www.cluecon.com

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org> 
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20161122/81ed9cf5/attachment.html