[Freeswitch-users] ReINVITE failure with ACK+SDP scenario
Shona McNeill
Shona.McNeill at Aculab.com
Tue Jul 2 20:13:40 MSD 2013
Try adding
<param name="enable-3pcc" value="proxy"/>
to the sip profile.
The horror that is "bypass media with proxy-mode 3pcc" should handle ACK+SDP after a no-SDP invite. :-)
At the least, it may point the way to what else needs to be added to make it all work.
Shona
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-
> users-bounces at lists.freeswitch.org] On Behalf Of BNML
> Sent: 02 July 2013 16:49
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] ReINVITE failure with ACK+SDP scenario
>
> I am suspecting a patch on mod_sofia migth be needed in order to
> resolve the Late SDP Negotiation in an RE-INVITE scenario to avoir the
> one-way audio. I can get involved in writing the patch, if needed.
>
> Here is the FS console logs along with sip trace
> http://pastebin.freeswitch.org/21141
>
> Le 01/07/2013 14:40, Muhammad Shahzad a écrit :
>
>
> Can you send FS console logs along with sip trace enabled? Just
> curious how FS responds to this situation.
>
> Thank you.
>
>
> On Mon, Jul 1, 2013 at 7:21 PM, BNML <bnml at andrexen.com> wrote:
>
>
> Hi,
>
> The problem is that FreeSWITCH doesn't forward Late SDP
> negotiation in a RE-INVITE scenario which end up to one way audio.
>
> After testing the option "enable-soa" didn't resolved the
> issue.
>
> I want to bridge two calls that are already established (I
> am not dealing with INVITE but with Re-INVITE) .
> That's why inbound-late-negotiation (which is within the
> dialplan) is not in the scope of this (I have it activated already).
>
>
> > When Call B is answered, you should use SDP in 200 OK as
> basis for SDP of ACK in Call A. Otherwise you will get one way audio as
> expect.
>
>
> Yes. This is the scenario I described at state 4) of the
> schema.
>
>
> Thanks
>
> BNML
>
> Le 06/28/2013 07:00 PM, Muhammad Shahzad a écrit :
>
>
> I am not sure what you are trying to do here. You
> said we want to originate two calls from your PABX to carrier through
> FS, i.e.
>
> Call A: PABX -> FS -> Carrier -> User A
> Call B: PABX -> FS -> Carrier -> User B
>
> And then bridge them on FS, such that,
>
> User A <- Carrier <- FS -> Carrier -> User B
>
> I think there are easier ways to achieve this. But
> anyway in this scenario,
>
> Since Call A does not have SDP, which means you are
> triggering Late SDP Negotiation. Make sure you have enabled in sofia
> profile,
>
> <param name="inbound-late-negotiation" value="true"/>
>
> So, that FS does not Answer it with local SDP.
> Additionally you may need to disable SOA,
>
> <param name="enable-soa" value="false"/>
>
> When 200 OK for Call A arrives (from carrier, through
> FS to PABX), you use it as basis for SDP of INVITE in Call B. That's
> interesting..
>
> When Call B is answered, you should use SDP in 200 OK
> as basis for SDP of ACK in Call A. Otherwise you will get one way audio
> as expect.
>
>
> Thank you.
>
>
>
>
> On Fri, Jun 28, 2013 at 6:56 PM, BNML
> <bnml at andrexen.com> wrote:
>
>
> Hi all,
>
> I am having trouble with a transfert failure in
> a special scenario which
> i wasn't able to find documentation or
> configuration option solving it.
>
> I have a PBX creating two outbound call, it
> want to bridge them on the
> FreeSWITCH.
> It is sending a first INVITE without SDP and it
> will send the SDP in the
> ACK.
>
> (This is not breaking RFC3261 but I admit
> rarely seen scenario.
> http://tools.ietf.org/html/rfc3261#section-
> 13.2.1 : "The UAC MAY choose
> to add a message body to the INVITE.". Looking
> at this scenario there
> is, indeed, at least one interesting scenario
> where it can helps
> avoiding creating another useless Re-INVITE
> transaction, see below)
>
> For the second call it is sending a re-invite
> with SDP.
> FreeSWITCH will understand and forward (because
> of "bypass_media") the
> second Re-INVITE but does never forward the
> first Re-INVITE (The one
> without SDP in the INVITE but in the ACK) to
> carrier, so this will end
> up in one-way-audio.
>
> A little schema to help :
>
> PBX FREESWITCH COMMENT
> 1) INVITE (a-leg) [THERE
> IS NO SDP]
> 2) 200 OK (fs-a-
> leg)[FreeSWITCH answer with the SDP
> information before any media update occurs, but
> my PBX is now aware of
> media informations]
> 3) INVITE+SDP (b-leg) [This
> is b-leg updated media
> information with previous 200 OK (fs-a-leg)]
> 4) 200 OK (fs-b-leg)[My PBX
> updated media and has
> b-leg freeswitch media information that it will
> send in the ACK+SDP to
> finalize the both-legs Re-INVITE]
> 5) ACK (b-leg)
> [Acknoledgement of b-leg
> Re-INVITE transaction]
> 6) ACK+SDP (a-leg) [ ***
> problem *** it assume
> FreeSWITCH will update media information and
> forward to the carrier
> which it won't]
>
> The solution i imagine is FreeSWITCH SHOULD
> trigger a Re-INVITE either
> for the a-leg at INVITE (without SDP in "1")
> receiption (and update
> media information when receiving the ACK+SDP)
> _OR_ create a Re-INVITE
> transaction at "6" when receiving ACK+SDP. One
> way or another, I am
> supposing FreeSWITCH refuse to forward the
> INVITE without SDP at first
> maybe because of carrier compatibility ? If
> not, this seems the way to go.
>
> Has anyone faced such a scenario ? Am I rigth
> assuming a patch is needed
> ? If yes, any suggestions are welcome.
>
> Thanks you
>
> BNML
>
>
>
> _________________________________________________________________
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>
> --
>
> Mit freundlichen Grüßen
> Muhammad Shahzad
> -----------------------------------
> CISCO Rich Media Communication Specialist (CRMCS)
> CISCO Certified Network Associate (CCNA)
> Cell: +49 176 99 83 10 85
> MSN: shari_786pk at hotmail.com
> Email: shaheryarkh at googlemail.com
>
>
>
>
> _________________________________________________________________
> ________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> FreeSWITCH-powered IP PBX: The CudaTel Communication
> Server
>
>
> 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
> http://www.freeswitch.org
> http://wiki.freeswitch.org
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>
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>
>
> --
>
> Mit freundlichen Grüßen
> Muhammad Shahzad
> -----------------------------------
> CISCO Rich Media Communication Specialist (CRMCS)
> CISCO Certified Network Associate (CCNA)
> Cell: +49 176 99 83 10 85
> MSN: shari_786pk at hotmail.com
> Email: shaheryarkh at googlemail.com
>
>
>
> _________________________________________________________________
> ________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
>
>
>
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