Sorry I meant : "I am _now_ able". Thanks you Le 02/07/2013 19:10, BNML a écrit :
I am not able to propagate the ACK+SDP (and the INVITE without SDP) to the carrier. Thanks !
Le mar. 02 juil. 2013 18:13:40 CEST, Shona McNeill a écrit :
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@lists.freeswitch.org [mailto:freeswitch- users-bounces@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@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@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@hotmail.com Email: shaheryarkh@googlemail.com
_________________________________________________________________ ________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
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Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
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_________________________________________________________________ ________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch- users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitc h-users http://www.freeswitch.org
--
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@hotmail.com Email: shaheryarkh@googlemail.com
_________________________________________________________________ ________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitc h-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org