[Freeswitch-users] absolute_codec_string not working

[Lợi Đặng]

In FS document of media proxy mode:
 > FreeSWITCH has no control or even understanding of other SDP parameters.
Look like I have to find another way, like writing a custom module
listening on specific event.
Any suggest?

Thanks to all of you.
rgds,

Loi Dang Thanh
Phone : 841224.735.448
Email : loi.dangthanh at gmail.com

On Wed, Nov 23, 2016 at 10:28 AM, Ken Rice <krice at freeswitch.org> wrote:

> 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> 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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> _________________________________________________________________________
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>
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