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@gmail.com 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= 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@gmail.com
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