<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 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 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 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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</div>
-- <br>
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<pre>_________________________________________________________________________
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<br></blockquote></div><br><br clear="all"><div><br></div>-- <br><div><span style="color:rgb(136,136,136);font-family:arial,sans-serif;font-size:13px;background-color:rgb(255,255,255)">Mit freundlichen Grüßen</span></div>
<span style="color:rgb(136,136,136);font-family:arial,sans-serif;font-size:13px;background-color:rgb(255,255,255)">Muhammad Shahzad</span><br style="color:rgb(136,136,136);font-family:arial,sans-serif;font-size:13px;background-color:rgb(255,255,255)">
<span style="color:rgb(136,136,136);font-family:arial,sans-serif;font-size:13px;background-color:rgb(255,255,255)">------------------------------</span><span style="color:rgb(136,136,136);font-family:arial,sans-serif;font-size:13px;background-color:rgb(255,255,255)">-----</span><br style="color:rgb(136,136,136);font-family:arial,sans-serif;font-size:13px;background-color:rgb(255,255,255)">
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