Thanks Michael,<br><br>I will try to find some solution/workaround.<br><br><div class="gmail_quote">2010/2/22 Michael Jerris <span dir="ltr"><<a href="mailto:mike@jerris.com">mike@jerris.com</a>></span><br><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
the port 0 with PT of 19 is sofia rejecting the sdp becuase we don't support it.<div><div></div><div class="h5"><br><br><div class="gmail_quote">On Mon, Feb 22, 2010 at 12:10 PM, Steve Underwood <span dir="ltr"><<a href="mailto:steveu@coppice.org" target="_blank">steveu@coppice.org</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">Hi Michael,<br>
<br>
On 02/23/2010 12:49 AM, ivdreg ivdreg wrote:<br>
> Hi Michael,<br>
><br>
> As I said in a previous mails I know exactly what is happening.<br>
> In working setup:<br>
><br>
> ITSP ---> YATE (GW - Frontend) ---> FreeSwitch (routing<br>
> server/xml_curl) ---> YATE (GW - SIP Interop) ---> OpenSIPS ---><br>
> Subscriber.<br>
><br>
> I prefer to change YATE (GW - Frontend) and YATE (GW - SIP Interop)<br>
> with FreeSwitch for some reasons. The problem is:<br>
><br>
> INVITE comes from ITSP and goes to subscriber via OpenSIPS. INIVITE<br>
> between FreeSwitch (routing server) and YATE (GW - SIP Interop)<br>
> contains SDP:<br>
<div>> m=audio 21108 RTP/AVP 18 4 8 0<br>
> c=IN IP4 10.10.1.110<br>
> a=rtpmap:18 G729/8000<br>
> a=rtpmap:4 G723/8000<br>
> a=rtpmap:8 PCMA/8000<br>
> a=rtpmap:0 PCMU/8000<br>
> m=image 21108 udptl t38<br>
> c=IN IP4 10.10.1.110<br>
> a=T38FaxVersion:0<br>
> a=T38MaxBitRate:14400<br>
> a=T38FaxUdpEC:t38UDPRedundancy<br>
> a=T38FaxRateManagement:<br>
> transferredTCF<br>
><br>
</div>> And reply 200 OK contains in SDP:<br>
> *m=audio 34788 RTP/AVP 8*<br>
<div>> a=rtpmap:8 PCMA/8000<br>
> a=silenceSupp:off - - - -<br>
> a=ptime:20<br>
><br>
</div>> Reply 200 OK SDP between YATE (GW - SIP Interop) and OpenSIPS contains<br>
> in SDP:<br>
> *m=audio 16330 RTP/AVP 8*<br>
<div>> a=rtpmap:8 PCMA/8000<br>
> a=silenceSupp:off - - - -<br>
> a=ptime:20<br>
</div>> *m=image 0 udptl 19*<br>
><br>
> In this case everything works fine. Line *m=image 0 udptl 19 *is<br>
> removed by YATE.<br>
> But same test with FreeSWITCH on the place of YATE (GW - SIP Interop)<br>
> *"m=image 0 udptl 19" *call brakes as you can see in my first mail.<br>
><br>
> I don't want to compare or discus YATE and FS functionality or<br>
> something else. I just see difference in behavior and because I'm not<br>
> a big expert don't know witch implementation is more accurate<br>
> according standards. And second: Is it impossible for me to upgrade<br>
> all CPE so only thing I can do is to fix it on server side. That is<br>
> because I ask for a help.<br>
You said the the broken line "m=image 0 udptl 19" goes from OpenSIPS to<br>
YATE. Do you know if it originates from the OpenSIPS box or the<br>
subscriber? If it originates from the OpenSIPS box it should be reported<br>
to them. If its from the subscriber, well...... your chances of getting<br>
anything fixed are usually small.<br>
<font color="#888888"><br>
Steve<br>
</font><div><div></div><div><br>
<br>
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