[Freeswitch-users] absolute_codec_string not working

[Lợi Đặng]

@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.
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?

rgds,

Loi Dang Thanh
Phone : +84.1224.735.448
Email : loi.dangthanh at gmail.com

On Tue, Nov 29, 2016 at 5:53 AM, Anthony Minessale <
anthony.minessale at 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 at 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 at gmail.com> wrote:
>>
>> 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 ...
>> I 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.
>>
>> rgds
>>
>> Loi Dang Thanh
>> Phone : 84.1224.735.448
>> Email : loi.dangthanh at gmail.com
>>
>> On Thu, Nov 24, 2016 at 12:33 AM, Michael Jerris <mike at 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 at gmail.com> wrote:
>>>
>>> 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=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}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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>
>
>
> --
> Anthony Minessale II       ♬ @anthmfs  ♬ @FreeSWITCH  ♬
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