[Freeswitch-users] absolute_codec_string not working

[Anthony Minessale]

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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