[Freeswitch-users] Codec negotiation with carriers
Anto
potxoka at gmail.com
Sat Mar 17 23:13:52 MSK 2012
Hello
I do not want to bother them, nor give me the solution. I would like
to learn how to configure it for myself (in fact one of the settings
works), but I do not understand the functioning correctly, for so not
to disturb in the future.
I attached the various configurations and their sip traces, as well as
the logs (debug):
First scenario ( Call isn´t established )
http://pastebin.freeswitch.org/18685
Second scenario ( Call isn´t established )
http://pastebin.freeswitch.org/18686
Third scenario ( Call isn´t established )
http://pastebin.freeswitch.org/18687
Fourth scenario ( Call isn´t established )
http://pastebin.freeswitch.org/18688
Fifth scenario ( Call established )
http://pastebin.freeswitch.org/18689
Sixth scenario ( Fail transcoding )
http://pastebin.freeswitch.org/18690
General settings:
vars.xml
<X-PRE-PROCESS cmd="set"
data="global_codec_prefs=VB32,G7221,speex,PCMU,PCMA,BV16,G726-32,iLBC,GSM,G729,G723,AMR"/>
<!--X-PRE-PROCESS cmd="set"
data="outbound_codec_prefs=PCMU,PCMA,G729,G723,AMR,VB32,G7221,speex,BV16,G726-32,iLBC,GSM"/-->
<X-PRE-PROCESS cmd="set"
data="carriers_codec_prefs=PCMU,PCMA,G729,G723,AMR,VB32,G7221,speex,BV16,G726-32,iLBC,GSM"/>
-----------------------------------------------------------------------------------------------------------------------------------------------
dialplan\outbound.xml
<include>
<context name="internal-in">
<extension name="outbound">
<condition field="network_addr" expression="^192\.168\.20\.40$" />
<condition field="destination_number" expression="^(\d+)$">
<action application="unset" data="sip_h_X-AUTH-IP"/>
<action application="set" data="inherit_codec=true"/>
<!--action application="set" data="proxy_media=true"/-->
<action application="set" data="hangup_after_bridge=true"/>
<!--action application="answer"/-->
<action application="bridge" data="sofia/gateway/carrier1-1/$1"/>
</condition>
</extension>
</context>
</include>
Thanks !
Regards
Anto
2012/3/16 Michael Collins <msc at freeswitch.org>:
> Well, this is a little better, however you don't have proper freeswitch logs
> on all these calls. For example, only the first call has freeswitch debug
> output. The other calls have sip traces, but not the first call. One call
> has what appears to be info-level output, but not debug-level output.
>
> I'd recommend that if you have this much information it might be good to put
> each call example in its own pastebin. Also, be sure to give a detailed
> description of what kind of call you are documenting. Some of your
> traces/debugs have no information explaining what the call is doing. Whether
> you are reporting a working or failed call, be sure to mention what kind of
> call it is. In the case of a failed call, be sure to mention what it is you
> are trying to do and what call result you expected to see.
>
> Thanks!
>
> -MC
>
>
> On Thu, Mar 15, 2012 at 2:22 PM, Anto <potxoka at gmail.com> wrote:
>>
>> Hi
>>
>> If, upload a file to trace and explanation to this address
>> http://pastebin.freeswitch.org/18599
>>
>> I do not want disturb watching this ;-), I prefer to use a system to
>> understand, this scenario and for future projects.
>>
>> With everything I've read do not really understand what the codecs :-S
>> , but if I had been good to understand the rest of the operation of
>> FreeSWITCH (or so I think). Thanks !
>>
>> Regards
>> Anto
>>
>> 2012/3/14 Michael Collins <msc at freeswitch.org>:
>> > Did you get sip traces and logs of working vs. non-working calls and put
>> > them on pastebin? Most likely there is an explanation but it will take
>> > some
>> > time and effort to figure it out.
>> >
>> > -MC
>> >
>> >
>> > On Wed, Mar 14, 2012 at 2:18 PM, Anto <potxoka at gmail.com> wrote:
>> >>
>> >> Hello
>> >>
>> >> I have searched previous messages in the list, I consulted the book of
>> >> FreeSWITCH (which I bought over a year), wiki and so on. I still do
>> >> not understand how and why in some cases I work. Also I downloaded
>> >> frontend to consult your code if there was something about this, but
>> >> still the same. I have several weeks with this question and I can not
>> >> find it. In the end I decided to spend the gateway to Asterisk, and
>> >> you at least understand its operation. Thank you very much to all :-)
>> >>
>> >> Best regards
>> >> Anto
>> >>
>> >> 2012/3/11 Anto <potxoka at gmail.com>:
>> >> > Hi
>> >> >
>> >> > I still do not find the solution and not really understanding,
>> >> > because
>> >> > it works:-S
>> >> >
>> >> > regards
>> >> > anto
>> >> >
>> >> > 2012/3/7 Anto <potxoka at gmail.com>:
>> >> >> Hello
>> >> >>
>> >> >> Attached file, with the traces of the different tests (with
>> >> >> different
>> >> >> configurations).
>> >> >>
>> >> >> http://pastebin.freeswitch.org/18599
>> >> >>
>> >> >> I have read the url that you mentioned, the initial guide
>> >> >> FreeSWITCH,
>> >> >> that of mod_sofia, applications, etc.. I believe that most of the
>> >> >> wiki
>> >> >> (maybe when do not give the solution, read as much documentation is
>> >> >> worse idea :-S, lock me more).
>> >> >>
>> >> >> I made a configuration that works (I have not tested the audio), but
>> >> >> earlier (before I started "touch" the profiles) if I could talk to a
>> >> >> physical phone (several times). The problem is that I can not
>> >> >> understand why it works and sometimes not, and I would like to learn
>> >> >> :-). Not only do and forget, so I would like to learn and less
>> >> >> disturbing to the mail list and (maybe in the future) to help other
>> >> >> newbies like me :-). Thanks !
>> >> >>
>> >> >> Best regards
>> >> >> Anto
>> >> >>
>> >> >> 2012/3/7 Michael Collins <msc at freeswitch.org>:
>> >> >>> You may want to read up on codec negotiation:
>> >> >>> http://wiki.freeswitch.org/wiki/Codec_negotiation
>> >> >>>
>> >> >>> There are different ways to handle codecs depending on your needs.
>> >> >>> I'd
>> >> >>> read
>> >> >>> that page first and then try out some of the suggestions. If you're
>> >> >>> still
>> >> >>> having trouble then I'd recommend getting SIP traces of the traffic
>> >> >>> and
>> >> >>> putting them on pastebin.freeswitch.org. The gang here is pretty
>> >> >>> good
>> >> >>> at
>> >> >>> looking over logs and helping with diagnosing problems. :)
>> >> >>>
>> >> >>> -MC
>> >> >>>
>> >> >>> On Tue, Mar 6, 2012 at 2:30 PM, Anto <potxoka at gmail.com> wrote:
>> >> >>>>
>> >> >>>> Hi
>> >> >>>>
>> >> >>>> I am following the steps in this direction
>> >> >>>> "http://wiki.freeswitch.org/wiki/SBC_Setup" and
>> >> >>>>
>> >> >>>> "http://wiki.freeswitch.org/wiki/Provider_Configuration:_Broadvoice",
>> >> >>>> I reread the whole entire wiki (or so I lack), but do not quite
>> >> >>>> assimilate or finding the right formula to operate the bridge :-S.
>> >> >>>>
>> >> >>>> I captured traffic with ngrep, I enabled sip-trace, console
>> >> >>>> logconsole
>> >> >>>> 8, etc., but unless the transcoding error (only two of the
>> >> >>>> hundreds
>> >> >>>> of
>> >> >>>> combinations of settings that I have), I have not seen anything
>> >> >>>> "weird" :-S
>> >> >>>>
>> >> >>>> I have 3 suppliers, each with this codec:
>> >> >>>>
>> >> >>>> 1) 2) 3)
>> >> >>>> G729 G729 G729
>> >> >>>> G711u G711A G711A
>> >> >>>> G711A G711u G711u
>> >> >>>> G723 G723
>> >> >>>> G722
>> >> >>>> GSM
>> >> >>>>
>> >> >>>> I think I understand that when making an outside call, FreeSWITCH
>> >> >>>> follow these steps:
>> >> >>>>
>> >> >>>> USER -> ( Dialplan -> profile (internal) -> bridge (external) ->
>> >> >>>> profile (external) ) -> PROVIDER
>> >> >>>>
>> >> >>>> PROVIDER -> ( Dialplan -> profile (external) -> bridge
>> >> >>>> (internal)
>> >> >>>> ->
>> >> >>>> profile (internal) ) -> USER
>> >> >>>>
>> >> >>>> right?
>> >> >>>>
>> >> >>>> Internal and external I set as follows (and not many changes have
>> >> >>>> done, and not remember it, because I've been testing days). If
>> >> >>>> outbound (outbound-codec-prefs) all codecs specified system does
>> >> >>>> not
>> >> >>>> handle the call, I have to specify these by hand. If active
>> >> >>>> inbound-proxy-media, not the caller. Some of the time I worked,
>> >> >>>> but
>> >> >>>> gave me an error that it can do transcoding G729 codec (I do
>> >> >>>> passthrough), but the proxy does not work half.
>> >> >>>>
>> >> >>>> If the outbound property (outbound-codec-prefs) all codecs
>> >> >>>> specified
>> >> >>>> system does not handle the call, I have to specify these by hand.
>> >> >>>> If
>> >> >>>> active inbound-proxy-media, not the caller. Some of the time I
>> >> >>>> worked,
>> >> >>>> but gave me an error that it can do transcoding G729 codec (I want
>> >> >>>> to
>> >> >>>> make passthrough), but the "proxy media" does not work.
>> >> >>>>
>> >> >>>> Basically, what I do is that local users can use all the codecs
>> >> >>>> allowed (iLBC, GSM, ...) and make an outside call, use the carrier
>> >> >>>> that will indicate the priority but the free codec.
>> >> >>>>
>> >> >>>> With this configuration works for me, but I would like to
>> >> >>>> understand
>> >> >>>> why so if it works and otherwise no. Coming to understand how to
>> >> >>>> configure properly and so as not to disturb the mail list ;-).
>> >> >>>> Thanks
>> >> >>>> !
>> >> >>>>
>> >> >>>> Best regards
>> >> >>>> Anto
>> >> >>>>
>> >> >>>> vars.xml
>> >> >>>>
>> >> >>>> <X-PRE-PROCESS cmd="set"
>> >> >>>>
>> >> >>>>
>> >> >>>>
>> >> >>>> data="global_codec_prefs=iLBC,G7221,speex,PCMU,PCMA,BV16,G726-32,GSM,G729,G723,AMR"/>
>> >> >>>> <X-PRE-PROCESS cmd="set"
>> >> >>>>
>> >> >>>>
>> >> >>>>
>> >> >>>> data="carriers_codec_prefs=PCMU,PCMA,G729,G723,AMR,iLBC,G7221,speex,BV16,G726-32,GSM"/>
>> >> >>>>
>> >> >>>> internal.xml
>> >> >>>>
>> >> >>>> <param name="codec-prefs" value="$${global_codec_prefs}"/>
>> >> >>>> <!--param name="inbound-codec-prefs"
>> >> >>>> value="$${global_codec_prefs}"/>
>> >> >>>> <param name="outbound-codec-prefs"
>> >> >>>> value="$${global_codec_prefs}"/-->
>> >> >>>> <!--param name="inbound-codec-negotiation" value="generous"/-->
>> >> >>>> <param name="inbound-codec-negotiation" value="greedy"/>
>> >> >>>> <param name="inbound-late-negotiation" value="true"/>
>> >> >>>> <param name="renegotiate-codec-on-reinvite" value="true"/>
>> >> >>>> <!--param name="inbound-proxy-media" value="true"/-->
>> >> >>>>
>> >> >>>> external.xml
>> >> >>>>
>> >> >>>> <param name="codec-prefs" value="$${carriers_codec_prefs}"/>
>> >> >>>> <param name="inbound-codec-prefs"
>> >> >>>> value="$${carriers_codec_prefs}"/>
>> >> >>>> <!--param name="outbound-codec-prefs"
>> >> >>>> value="$${carriers_codec_prefs}"/-->
>> >> >>>> <param name="outbound-codec-prefs" value="PCMU,PCMA,G729,G723"/>
>> >> >>>> <!--param name="inbound-codec-negotiation" value="generous"/-->
>> >> >>>> <param name="inbound-codec-negotiation" value="greedy"/>
>> >> >>>> <param name="inbound-late-negotiation" value="true"/>
>> >> >>>> <param name="renegotiate-codec-on-reinvite" value="true"/>
>> >> >>>> <!--param name="inbound-proxy-media" value="true"/-->
>> >> >>>>
>> >> >>>> dialplan/outbound.xml
>> >> >>>>
>> >> >>>> <include>
>> >> >>>> <context name="internal">
>> >> >>>> <extension name="outbound">
>> >> >>>> <condition field="destination_number"
>> >> >>>> expression="^(\d+)$">
>> >> >>>> <!--action application="set"
>> >> >>>> data="inherit_codec=true"/-->
>> >> >>>> <!--action application="set"
>> >> >>>> data="hangup_after_bridge=true"/-->
>> >> >>>> <!--action application="set"
>> >> >>>> data="proxy_media=true"/-->
>> >> >>>> <action application="bridge"
>> >> >>>> data="sofia/gateway/provider-2/$1"/>
>> >> >>>> </condition>
>> >> >>>> </extension>
>> >> >>>> </context>
>> >> >>>>
>> >> >>>>
>> >> >>>>
>> >> >>>> _________________________________________________________________________
>> >> >>>> Professional FreeSWITCH Consulting Services:
>> >> >>>> consulting at freeswitch.org
>> >> >>>> http://www.freeswitchsolutions.com
>> >> >>>>
>> >> >>>>
>> >> >>>>
>> >> >>>>
>> >> >>>> Official FreeSWITCH Sites
>> >> >>>> http://www.freeswitch.org
>> >> >>>> http://wiki.freeswitch.org
>> >> >>>> http://www.cluecon.com
>> >> >>>>
>> >> >>>> FreeSWITCH-users mailing list
>> >> >>>> FreeSWITCH-users at lists.freeswitch.org
>> >> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >>>>
>> >> >>>>
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>> >> >>>> http://www.freeswitch.org
>> >> >>>
>> >> >>>
>> >> >>>
>> >> >>>
>> >> >>>
>> >> >>> _________________________________________________________________________
>> >> >>> Professional FreeSWITCH Consulting Services:
>> >> >>> consulting at freeswitch.org
>> >> >>> http://www.freeswitchsolutions.com
>> >> >>>
>> >> >>>
>> >> >>>
>> >> >>>
>> >> >>> Official FreeSWITCH Sites
>> >> >>> http://www.freeswitch.org
>> >> >>> http://wiki.freeswitch.org
>> >> >>> http://www.cluecon.com
>> >> >>>
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>> >> >>>
>> >>
>> >>
>> >> _________________________________________________________________________
>> >> Professional FreeSWITCH Consulting Services:
>> >> consulting at freeswitch.org
>> >> http://www.freeswitchsolutions.com
>> >>
>> >>
>> >>
>> >>
>> >> Official FreeSWITCH Sites
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>> >>
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>> >
>> >
>> >
>> >
>> > _________________________________________________________________________
>> > Professional FreeSWITCH Consulting Services:
>> > consulting at freeswitch.org
>> > http://www.freeswitchsolutions.com
>> >
>> >
>> >
>> >
>> > Official FreeSWITCH Sites
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>> > http://wiki.freeswitch.org
>> > http://www.cluecon.com
>> >
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>>
>> _________________________________________________________________________
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>> http://www.freeswitchsolutions.com
>>
>>
>>
>>
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>
>
>
> _________________________________________________________________________
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> http://www.freeswitchsolutions.com
>
>
>
>
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