[Freeswitch-users] Freeswitch transcoding

[Sidharth Kshatriya]

Nice answer!

On Tue, Jun 14, 2011 at 6:44 PM, Steven Ayre <steveayre at gmail.com> wrote:

> At a guess, it's this line:
>
> <action application="set" data="bypass_media=true"/>
> Unless the media is going through FS it can't trancode. The SDP (codec
> list) will be forwarded from the aleg to the bleg so absolute_codec_string
> would be ignored.
>
> Also this has no effect, unless have proxy_media set to true on the sip
> profile. You don't need to set them to false, they'll already be false.
> You're correct to have this disabled though, as that would also prevent
> transcoding.
>
> <action application="set" data="proxy_media=false"/>
>
> This won't do as you think:
>
> <action application="set" data="absolute_codec_string=PCMU"/>
> It chooses the codec before you get to the dialplan. absolute_codec_string
> will only apply to the bleg. The incoming codec will either be picked from
> the inbound-codec-prefs param on the sip profile that receives the a-leg, or
> if you're using late-negotiation it'll pick that after the bridge so that
> it'll try to use the same codec as the bleg if possible and pick one of the
> other choices and transcode if that's not possible. absolute_codec_string
> overrides the outbound-codec-prefs sip profile param on the bleg.
>
> Try these on your sip profile:
>  <param name="inbound-codec-prefs" value="PCMU" />
>  <param name="outbound-codec-prefs" value="PCMA" />
>
> Then this will transcode:
>
> <extension name="external">
>      <condition field="destination_number" expression="^666654351428063">
>           <action application="set" data="hangup_after_bridge=true"/>
>           <action application="set" data="progress_timeout=20"/>
>           <action application="set"
> data="export_vars=et_routing_number,et_routing_host,et_routed_number,et_routed_host,et_random_key"/>
>           <action application="set"
> data="et_routing_number=666654351428063"/>
>           <action application="set" data="et_routing_host=10.0.0.12"/>
>           <action application="set"
> data="et_random_key=029645322327599477"/>
>           <action application="set"
> data="et_routed_number=555554351428063"/>
>           <action application="set" data="et_routed_host=10.0.0.14"/>
>           <action application="bridge" data="sofia/external/
> 555554351428063 at 10.0.0.14"/>
>      </condition>
> </extension>
>
> If you want to offer other codecs you can add them to the codec-pref
> params. They'll be listed in the preference order. FS will handle the
> transcoding automatically.
>
> You might also find it useful to read the Codec Negotiation page on the
> Wiki, if you haven't already done so:
> http://wiki.freeswitch.org/wiki/Codec_Negotiation
>
> -Steve
>
>
>
>
>
> On 14 June 2011 13:35, Gustavo Espeche <gustavo.espeche at easyipcall.com>wrote:
>
>> Hi we are trying to do transcoding with freeswitch g711a-->g711u or
>> g711-->speex but we can't do that FS do a trascoding, follow is a
>> dialplan that we use with mod_curl
>>
>> <?xml version="1.0" encoding="UTF-8" standalone="no"?>
>> <document type="freeswitch/xml">
>> <section name="dialplan" description="Regex/XML Dialplan">
>> <context name="public">
>> <extension name="external">
>> <condition field="destination_number" expression="^666654351428063">
>> <action application="set" data="hangup_after_bridge=true"/>
>> <action application="set" data="progress_timeout=20"/>
>> <action application="set"
>>
>> data="export_vars=et_routing_number,et_routing_host,et_routed_number,et_routed_host,et_random_key"/>
>> <action application="set" data="et_routing_number=666654351428063"/>
>> <action application="set" data="et_routing_host=10.0.0.12"/>
>> <action application="set" data="et_random_key=029645322327599477"/>
>> <action application="set" data="proxy_media=false"/>
>> <action application="set" data="bypass_media=true"/>
>> <action application="set" data="absolute_codec_string=PCMU"/>
>> <action application="export" data="nolocal:absolute_codec_string=PCMA"/>
>> <action application="set" data="et_routed_number=555554351428063"/>
>> <action application="set" data="et_routed_host=10.0.0.14"/>
>> <action application="bridge"
>> data="sofia/external/555554351428063 at 10.0.0.14"/>
>> </condition>
>> </extension>
>> </context>
>> </section>
>> </document>
>>
>>  we relay can't understand what are we doing wrong.
>> are we need configure someone special in FS? we are using
>> sip_external_profile with port 5060.
>>
>>
>> Regards.
>>
>> --
>>
>> Gustavo Espeche
>> EasyIpCall S.R.L.
>> www.easyipcall.com
>> Bv Mitre 517 24° E
>> Cordoba - Argentina
>> Te: +54 - 351 - 4280633
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Sidharth Kshatriya
www.sidk.info
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110614/173fe7b6/attachment-0001.html