[Freeswitch-users] absolute_codec_string not working

[Lợi Đặng]

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> 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> 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
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