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<div class="moz-cite-prefix">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.<br>
<br>
Here is the FS console logs along with sip trace <br>
<a class="moz-txt-link-freetext" href="http://pastebin.freeswitch.org/21141">http://pastebin.freeswitch.org/21141</a><br>
<br>
Le 01/07/2013 14:40, Muhammad Shahzad a écrit :<br>
</div>
<blockquote
cite="mid:CAFZQphxAQjpYT8jXMkKb0g6pX5Fm=F+4HnCULNNCSajZcFcEXw@mail.gmail.com"
type="cite">
<div dir="ltr">Can you send FS console logs along with sip trace
enabled? Just curious how FS responds to this situation.
<div><br>
</div>
<div>Thank you.</div>
</div>
<div class="gmail_extra"><br>
<br>
<div class="gmail_quote">On Mon, Jul 1, 2013 at 7:21 PM, BNML <span
dir="ltr"><<a moz-do-not-send="true"
href="mailto:bnml@andrexen.com" target="_blank">bnml@andrexen.com</a>></span>
wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0
.8ex;border-left:1px #ccc solid;padding-left:1ex">
<div bgcolor="#FFFFFF" text="#000000">
<div>
<div class="im">Hi,<br>
<br>
The problem is that FreeSWITCH doesn't forward Late
SDP negotiation in a RE-INVITE scenario which end up
to one way audio.<br>
<br>
After testing the option "enable-soa" didn't resolved
the issue.<br>
<br>
I want to bridge two calls that are already
established (I am not dealing with INVITE but with
Re-INVITE) .<br>
That's why inbound-late-negotiation (which is within
the dialplan) is not in the scope of this (I have it
activated already).<br>
<br>
</div>
<div>
<div class="im">> 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.<br>
<br>
</div>
<div class="im"> Yes. This is the scenario I described
at state 4) of the schema.<br>
</div>
</div>
<div class="im"> <br>
Thanks<br>
<br>
BNML<br>
<br>
Le 06/28/2013 07:00 PM, Muhammad Shahzad a écrit :<br>
</div>
</div>
<blockquote type="cite">
<div dir="ltr">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.
<div>
<div class="h5">
<div><br>
</div>
<div>Call A: PABX -> FS -> Carrier ->
User A</div>
<div>Call B: PABX -> FS -> Carrier ->
User B</div>
<div><br>
</div>
<div>And then bridge them on FS, such that,</div>
<div><br>
</div>
<div>User A <- Carrier <- FS -> Carrier
-> User B</div>
<div><br>
</div>
<div>I think there are easier ways to achieve
this. But anyway in this scenario,</div>
<div><br>
</div>
<div>Since Call A does not have SDP, which means
you are triggering Late SDP Negotiation. Make
sure you have enabled in sofia profile,</div>
<div><br>
</div>
<div>
<pre style="line-height:1.1em;background-color:rgb(249,249,249);border:1px dashed rgb(47,111,171);padding:1em"><param name="inbound-late-negotiation" value="true"/></pre>
</div>
<div><br>
</div>
<div>So, that FS does not Answer it with local
SDP. Additionally you may need to disable SOA,</div>
<div><br>
</div>
<div>
<pre style="line-height:1.1em;background-color:rgb(249,249,249);border:1px dashed rgb(47,111,171);padding:1em"><param name="enable-soa" value="false"/></pre>
</div>
<div><br>
</div>
<div>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..</div>
<div><br>
</div>
<div>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.</div>
<div><br>
</div>
<div><br>
</div>
<div> Thank you.</div>
<div><br>
</div>
<div><br>
</div>
</div>
</div>
</div>
<div>
<div class="h5">
<div class="gmail_extra"><br>
<br>
<div class="gmail_quote">On Fri, Jun 28, 2013 at
6:56 PM, BNML <span dir="ltr"><<a
moz-do-not-send="true"
href="mailto:bnml@andrexen.com"
target="_blank">bnml@andrexen.com</a>></span>
wrote:<br>
<blockquote class="gmail_quote" style="margin:0
0 0 .8ex;border-left:1px #ccc
solid;padding-left:1ex">Hi all,<br>
<br>
I am having trouble with a transfert failure
in a special scenario which<br>
i wasn't able to find documentation or
configuration option solving it.<br>
<br>
I have a PBX creating two outbound call, it
want to bridge them on the<br>
FreeSWITCH.<br>
It is sending a first INVITE without SDP and
it will send the SDP in the<br>
ACK.<br>
<br>
(This is not breaking RFC3261 but I admit
rarely seen scenario.<br>
<a moz-do-not-send="true"
href="http://tools.ietf.org/html/rfc3261#section-13.2.1"
target="_blank">http://tools.ietf.org/html/rfc3261#section-13.2.1</a>
: "The UAC MAY choose<br>
to add a message body to the INVITE.". Looking
at this scenario there<br>
is, indeed, at least one interesting scenario
where it can helps<br>
avoiding creating another useless Re-INVITE
transaction, see below)<br>
<br>
For the second call it is sending a re-invite
with SDP.<br>
FreeSWITCH will understand and forward
(because of "bypass_media") the<br>
second Re-INVITE but does never forward the
first Re-INVITE (The one<br>
without SDP in the INVITE but in the ACK) to
carrier, so this will end<br>
up in one-way-audio.<br>
<br>
A little schema to help :<br>
<br>
PBX FREESWITCH
COMMENT<br>
1) INVITE (a-leg) [THERE
IS NO SDP]<br>
2) 200 OK
(fs-a-leg)[FreeSWITCH answer with the SDP<br>
information before any media update occurs,
but my PBX is now aware of<br>
media informations]<br>
3) INVITE+SDP (b-leg) [This
is b-leg updated media<br>
information with previous 200 OK (fs-a-leg)]<br>
4) 200 OK (fs-b-leg)[My
PBX updated media and has<br>
b-leg freeswitch media information that it
will send in the ACK+SDP to<br>
finalize the both-legs Re-INVITE]<br>
5) ACK (b-leg)
[Acknoledgement of b-leg<br>
Re-INVITE transaction]<br>
6) ACK+SDP (a-leg) [ ***
problem *** it assume<br>
FreeSWITCH will update media information and
forward to the carrier<br>
which it won't]<br>
<br>
The solution i imagine is FreeSWITCH SHOULD
trigger a Re-INVITE either<br>
for the a-leg at INVITE (without SDP in "1")
receiption (and update<br>
media information when receiving the ACK+SDP)
_OR_ create a Re-INVITE<br>
transaction at "6" when receiving ACK+SDP. One
way or another, I am<br>
supposing FreeSWITCH refuse to forward the
INVITE without SDP at first<br>
maybe because of carrier compatibility ? If
not, this seems the way to go.<br>
<br>
Has anyone faced such a scenario ? Am I rigth
assuming a patch is needed<br>
? If yes, any suggestions are welcome.<br>
<br>
Thanks you<br>
<br>
BNML<br>
<br>
<br>
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<pre wrap="">_________________________________________________________________________
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