<div dir="ltr">They said the 1st carrier rejected it so the second 183 is from a second carrier.. but yes, the first carrier shouldn't be sending a 183...<div>Anyway -- if you <i>please</i> look at the pcap from the first email, the log I posted a link to later, or the past two emails -- you'll see FS originally offers both PCMU and PCMA in the initial invite.</div>
<div><br></div><div>I think we need Anthony to chime in here... he knows the RFC stuff!</div><div><br clear="all"><div dir="ltr"><span style="font-family:Verdana,Arial,Helvetica,sans-serif"><span style="font-size:small">-Avi</span></span></div>
<br>
<br><br><div class="gmail_quote">On Tue, Jul 3, 2012 at 6:22 PM, Kristian Kielhofner <span dir="ltr"><<a href="mailto:kris@kriskinc.com" target="_blank">kris@kriskinc.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
That's pretty nasty of voxbeam to do... Considering they return the<br>
first 183 less than one second after the initial INVITE I'd say it's<br>
very likely they're providing false ringback and/or FAS (false answer<br>
supervision). Meanwhile the actual ringback comes over two seconds<br>
later with PCMA. That's shady and if I were you I'd switch carriers<br>
just on principle.<br>
<br>
However, as I said before they're not doing anything that's not<br>
allowed by the various specifications although FAS is considered by<br>
many as fraud.<br>
<br>
What are your codec settings like? Perhaps FS is rejecting PCMA<br>
because you're not allowing it?<br>
<div class="HOEnZb"><div class="h5"><br>
On Tue, Jul 3, 2012 at 9:38 AM, Avi Marcus <<a href="mailto:avi@avimarcus.net">avi@avimarcus.net</a>> wrote:<br>
> If so.. which I think is not.. is this a bug then in FS?<br>
><br>
> <a href="http://pastebin.freeswitch.org/19423" target="_blank">http://pastebin.freeswitch.org/19423</a><br>
> You see from the first SDP that PCMU and PCMA were both options.<br>
> Then for the next SDP FreeSWITCH was only considering PCMU which was the one<br>
> that got chosen by the first SDP.<br>
><br>
> Voxbeam claims it's not their fault.. is it?<br>
><br>
> -Avi<br>
><br>
><br>
><br>
> On Tue, Jul 3, 2012 at 4:26 PM, Kristian Kielhofner <<a href="mailto:kris@kriskinc.com">kris@kriskinc.com</a>><br>
> wrote:<br>
>><br>
>> Avi,<br>
>><br>
>> Multiple 18x responses that change the SDP are allowed. I can't<br>
>> find the specific document text now but as a random guy on the<br>
>> internet (for whatever that's worth) I'm certain it is allowed.<br>
>><br>
>> A re-INVITE can't be sent from either side until the dialog has been<br>
>> established (200+ACK). In a case where the UAC (caller) would like to<br>
>> update the session before it is established method UPDATE must be<br>
>> used.<br>
>><br>
>> On Tue, Jul 3, 2012 at 1:04 AM, Avi Marcus <<a href="mailto:avi@avimarcus.net">avi@avimarcus.net</a>> wrote:<br>
>> > Are multiple 183s from the endpoint that changes the SDP allowed? I'm<br>
>> > under<br>
>> > the impression this is broken, similar to<br>
>> > <a href="http://wiki.freeswitch.org/wiki/NDLB#NDLB-allow-nondup-sdp" target="_blank">http://wiki.freeswitch.org/wiki/NDLB#NDLB-allow-nondup-sdp</a><br>
>> > ... which is why when the codec changes, FS freaks out and cancels the<br>
>> > call<br>
>> > because of codec negotiation error.<br>
>> > Does that NDLB flag allow this, too?<br>
>> ><br>
>> > First has: audio 10116 RTP/AVP 0 101 13<br>
>> > Second has: audio 49020 RTP/AVP 8 13 101<br>
>> > PCAP: <a href="http://ge.tt/7MpyBwJ" target="_blank">http://ge.tt/7MpyBwJ</a><br>
>> ><br>
>> > Can someone point me to the specific RFC so I can tell the supplier to<br>
>> > fix<br>
>> > it?<br>
>> > And just curious.. what would make this allowed? A re-INVITE..?<br>
>> ><br>
>> > -Avi<br>
>> ><br>
>> ><br>
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>> --<br>
>> Kristian Kielhofner<br>
>><br>
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<br>
<br>
--<br>
Kristian Kielhofner<br>
<br>
_________________________________________________________________________<br>
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</div></div></blockquote></div><br></div></div>