On Nov 28, 2016, at 9:57 PM, Lợi Đặng <loi.dangthanh@gmail.com> wrote:_________________________________________________________________________rgds,Anyway, I'm still not sure why I shouldn't be using media proxy? since I have another point in my voice stack for codec negotiation?@Anthony, in inbound-proxy-media="true" in sip profile conf, I don't think setting absolute_codec_string can handle codec list anymore. I tried and it doesn't work.@Michael, referred to your link, I've done what I'm trying to accomplish, by setting switch_r_sdp variable, so many thanks.On Tue, Nov 29, 2016 at 5:53 AM, Anthony Minessale <anthony.minessale@gmail.com> wrote:I answered the question about the escaped comma 6 days ago. That is the answer to why doesn't absolute_codec_string work...On Mon, Nov 28, 2016 at 10:09 AM, Michael Jerris <mike@jerris.com> wrote:______________________________FreeSWITCH knows about g729 just fine. you shouldn’t be using proxy. Check out the page i referenced below for information on how to accomplish it, Proxy media mode is not the way you want to use for sure.On Nov 23, 2016, at 11:35 PM, Lợi Đặng <loi.dangthanh@gmail.com> wrote:rgdsI want to limit codecs choice for each callee accordingly, instead of fully pass `G729,PCMA,PCMU,GSM` to every callee, so that they don't know my full supported codec.actually, my FS needs to be in proxy_media mode, since it always deal with codecs it doesn't know about, g729.real case: my caller(asterisk) always compose INVITE with `G729,PCMA,PCMU,GSM` to FS, some of my callee only accept G729, while others accept G729,PCMA, and so on ...On Thu, Nov 24, 2016 at 12:33 AM, Michael Jerris <mike@jerris.com> wrote:I’m not totally sure what you are trying to accomplish but proxy_media is completely unnecessary and undesired for what you are doing. It should ONLY be used in the case where you are trying to pass codecs we don’t know about at all. Take a look at the codec negotiation page on freeswitch.org/confluence and I think you will find your answers. I don’t think you need a custom mod looking for events with what you have described so far.On Nov 23, 2016, at 5:37 AM, Lợi Đặng <loi.dangthanh@gmail.com> wrote:Any suggest?In FS document of media proxy mode:Look like I have to find another way, like writing a custom module listening on specific event.
> FreeSWITCH has no control or even understanding of other SDP parameters.Thanks to all of you.rgds,On Wed, Nov 23, 2016 at 10:28 AM, Ken Rice <krice@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@lists
.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org ] On Behalf Of L?i Ð?ng
Sent: Tuesday, November 22, 2016 9:07 PM
To: FreeSWITCH Users Help <freeswitch-users@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@gmail.com
On Tue, Nov 22, 2016 at 9:57 PM, Michael Jerris <mike@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@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=P
CMU,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}so
fia/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@gmail.com
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