[Freeswitch-users] inherit_codec failing with pre-media

sangdrax8 sangdrax8 at gmail.com
Wed Sep 25 18:15:57 MSD 2013 

Thank you for the explanation!  I went back and double checked, and
with freeswitch's ring commented out I do in fact get the 180's as you
described.  In fact, the ringing sound is there as well, it was just
providing a much quieter beep rather than a ring, so I must have
missed it.

So this makes sense then.  As long as you don't ask freeswitch to send
a ring of a specific type, then inherit_codec can work.  I'll have to
look at my sip client and see if I can configure this ring to sound
like a standard ring my users would expect.

thank you, I'll continue to work on this!

On Mon, Sep 23, 2013 at 10:33 AM, Steven Ayre <steveayre at gmail.com> wrote:
>> No, and I assumed this to be normal?  When one sip client places a
>> call to a second sip client on the same Freeswitch, I wouldn't expect
>> the second sip client to provide some sort of ringing.  In fact,
>> shouldn't this ringing be happening while the B-leg is being
>> contacted?  I don't see how it could be the one providing the sound.
>> In the example dialplan Freeswitch does provide this ringback, but in
>> doing so it also negotiates a codec.  Hence I removed the ringback,
>> and suddenly my inherit_codec is providing the functionality expected
>> minus the ring.
>
>
> Usually in SIP calling A -> FS -> B
>
> Bridge sends INVITE to B. 100 Trying is sent back immediately to
> acknowledge. A hears nothing.
>
> Then you receive either 180 or 183/SDP:
> - 180 Ringing indicates the phone is ringing but that A should generate its
> own ringback tone
> - 183/SDP indicates that B is sending early media (usually containing a
> ringback tone), FS forwards this to A
>
> Then you get 200 OK (answer) at which point you get 2-way audio setup.
>
> So either A or B generates the ringback, FS usually does not need to.
>
>
>
>
>
>
>
>
>
>
>
>
> On 23 September 2013 12:55, sangdrax8 <sangdrax8 at gmail.com> wrote:
>>
>> > Just to clarify this message, that's the 'set' application logging that
>> > the
>> > variable has been set on the channel, not anything suggesting that it is
>> > taking effect.
>>
>> Correct, the variable has been set.
>>
>> > Is the b-leg of the bridge not sending any early media ringback?
>>
>> No, and I assumed this to be normal?  When one sip client places a
>> call to a second sip client on the same Freeswitch, I wouldn't expect
>> the second sip client to provide some sort of ringing.  In fact,
>> shouldn't this ringing be happening while the B-leg is being
>> contacted?  I don't see how it could be the one providing the sound.
>> In the example dialplan Freeswitch does provide this ringback, but in
>> doing so it also negotiates a codec.  Hence I removed the ringback,
>> and suddenly my inherit_codec is providing the functionality expected
>> minus the ring.
>>
>>
>> On Thu, Sep 19, 2013 at 11:06 AM, Steven Ayre <steveayre at gmail.com> wrote:
>> >> It logs that it is set to true
>> >
>> >
>> > Just to clarify this message, that's the 'set' application logging that
>> > the
>> > variable has been set on the channel, not anything suggesting that it is
>> > taking effect.
>> >
>> >> I just can't have a phone system with no ringing for the calling party.
>> >
>> >
>> > Is the b-leg of the bridge not sending any early media ringback?
>> >
>> >
>> > On 19 September 2013 13:03, sangdrax8 <sangdrax8 at gmail.com> wrote:
>> >>
>> >> Yes, I did make sure that was activated in the profile.  the
>> >> configuration that comes from the git repo has that turned on, and I
>> >> am using the demo config directly from git.
>> >>
>> >> The demo dial plan doesn't include the inherit_codec=true line, so I
>> >> did add that.  It logs that it is set to true, but freeswitch ends up
>> >> transcoding.  If I remove the ringback, suddenly everything works as
>> >> expected.  I just can't have a phone system with no ringing for the
>> >> calling party.  Still it does indicate that inbound late negotiation
>> >> and the inherit_codec are capable of doing the right thing.  Just when
>> >> a ring back is added, it seems to fail to work.
>> >>
>> >> If I am configuring this correctly (only 1 change from the demo
>> >> configuration provided) should I open a bug somewhere, or e-mail a dev
>> >> list?  I didn't want to do either of these without someone confirming
>> >> that my configuration part is correct and that it is a bug.
>> >>
>> >> "Faithless is he, who says 'farewell', when the path darkens."
>> >> "you just keep on trying till you run out of cake"
>> >>
>> >>
>> >> On Wed, Sep 18, 2013 at 7:31 PM, Steven Ayre <steveayre at gmail.com>
>> >> wrote:
>> >> > Inherit_codec requires late negotiation - is that enabled too?
>> >> > Otherwise
>> >> > the
>> >> > aleg codec has already been picked before hitting the dialplan.
>> >> >
>> >> >
>> >> > On Wednesday, September 18, 2013, sangdrax8 wrote:
>> >> >>
>> >> >> Did I explain a bit to much detail for the users list?
>> >> >>
>> >> >> Does anyone use "inherit_codec=true" and have it working?
>> >> >>
>> >> >>
>> >> >> On Tue, Sep 17, 2013 at 2:27 PM, sangdrax8 <sangdrax8 at gmail.com>
>> >> >> wrote:
>> >> >> > I pulled the latest freeswitch from git this morning, and am still
>> >> >> > having issues using inherit_codec=true.  It seems the only way I
>> >> >> > can
>> >> >> > get it to defer to the choice of the b-leg call is to remove the
>> >> >> > ringback, and thus the need for early media.
>> >> >> >
>> >> >> > My setup has 2 phones with the following codecs configured:
>> >> >> >
>> >> >> > phone1000:  g722, ulaw
>> >> >> > phone1001:  ulaw
>> >> >> >
>> >> >> > My goal is to have a call initiated by phone1000 (a-leg) to defer
>> >> >> > the
>> >> >> > codec choice until phone1001 (b-leg) chooses one.  This would
>> >> >> > allow
>> >> >> > them to talk with no transcoding required by freeswitch.  With the
>> >> >> > following dial plan, this fails to work.
>> >> >> >
>> >> >> > <extension name="Local_Extension">
>> >> >> >       <condition field="destination_number"
>> >> >> > expression="^(10[01][0-9])$">
>> >> >> >         <action application="export" data="dialed_extension=$1"/>
>> >> >> >         <action application="set" data="ringback=${us-ring}"/>
>> >> >> >         <action application="set"
>> >> >> > data="transfer_ringback=$${hold_music}"/>
>> >> >> >         <action application="set" data="call_timeout=30"/>
>> >> >> >         <action application="set"
>> >> >> > data="hangup_after_bridge=true"/>
>> >> >> >         <action application="set" data="inherit_codec=true" />
>> >> >> >         <action application="bridge"
>> >> >> > data="user/${dialed_extension}@${domain_name}"/>
>> >> >> >       </condition>
>> >> >> > </extension>
>> >> >> >
>> >> >> > It does log that it is going to defer:
>> >> >> >
>> >> >> > 2013-09-17 13:15:10.956535 [DEBUG] mod_sofia.c:4542
>> >> >> > [zrtp_passthru]
>> >> >> > Setting a-leg inherit_codec=true
>> >> >> > 2013-09-17 13:15:10.956535 [DEBUG] mod_sofia.c:4545
>> >> >> > [zrtp_passthru]
>> >> >> > Setting b-leg
>> >> >> >
>> >> >> > absolute_codec_string='G722 at 8000h@20i at 64000b,PCMU at 8000h@20i at 64000b'
>> >> >> >
>> >> >> >
>> >> >> > then it tries to provide the ringback for the a-leg, and it
>> >> >> > negotiates
>> >> >> > to g722 since freeswitch supports that and it is the first match.
>> >> >> > Later when the b-leg is answered, it also negotiates, this time to
>> >> >> > ulaw as that is all b supports.  My call completes, but freeswitch
>> >> >> > has
>> >> >> > to transcode between the two.
>> >> >> >
>> >> >> > In an attempt to see inherit_codec work, I then commented out the
>> >> >> > ringback line from the above dialplan.  With only this one change,
>> >> >> > freeswitch now waits until the b-leg negotiation, and passes that
>> >> >> > back
>> >> >> > to the a-leg.  The call completes and both handsets are using ulaw
>> >> >> > and
>> >> >> > no transcoding is required.  The problem being that the calling
>> >> >> > party
>> >> >> > never gets a ring while waiting.
>> >> >> >
>> >> >> > I can't imagine that the only way to use inherit_codec=true is
>> >> >> > with
>> >> >> > all ringing turned off.  Can someone tell me if there is an
>> >> >> > additional
>> >> >> > option I am missing?  All documentation I read seems to indicate
>> >> >> > that
>> >> >> > just adding "inherit_codec=true" will fix things, but even the
>> >> >> > default.xml provided as an example uses ringback and therefore
>> >> >> > fails.
>> >> >>
>> >> >>
>> >> >>
>> >> >> _________________________________________________________________________
>> >> >> Professional FreeSWITCH Consulting Services:
>> >> >> consulting at freeswitch.org
>> >> >> http://www.freeswitchsolutions.com
>> >> >>
>> >> >> 
>> >> >> 
>> >> >>
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>> >> >>
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>> >> >
>> >> >
>> >> >
>> >> >
>> >> > _________________________________________________________________________
>> >> > 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
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>> >> >
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>> >> >
>> >>
>> >>
>> >> _________________________________________________________________________
>> >> 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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>> >>
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>> >
>> >
>> >
>> >
>> > _________________________________________________________________________
>> > 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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>>
>> _________________________________________________________________________
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>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
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>>
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>
>
>
> _________________________________________________________________________
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> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
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>
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