[Freeswitch-users] Codec negotiation with carriers

Michael Collins msc at freeswitch.org
Thu Mar 15 00:24:43 MSK 2012


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>
> >>>>
> >>>>
> _________________________________________________________________________
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> >>>> http://www.freeswitchsolutions.com
> >>>>
> >>>> 
> >>>> 
> >>>>
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> >>>
> >>>
> >>>
> >>>
> _________________________________________________________________________
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> >>> http://www.freeswitchsolutions.com
> >>>
> >>> 
> >>> 
> >>>
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> http://www.freeswitchsolutions.com
>
> 
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