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Propably you refer RFC 2327:<br>
<pre class="newpage"> Note that RTP audio formats typically do not include information
about the number of samples per packet. If a non-default (as
defined in the RTP Audio/Video Profile) packetisation is required,
the "ptime" attribute is used as given below.
or RFC 4566:
Note: RTP audio formats typically do not include information
about the number of samples per packet. If a non-default (as
defined in the RTP Audio/Video Profile) packetisation is
required, the "ptime" attribute is used as given above.
</pre>
IHMO that means, that the default packetisation CAN be used, if
nothing is specified, but it doesn't mean that it SHOULD be used.
But even if a ptime is specified, the RFC only says that the
specified ptime SHOULD be used, what is different from that it HAVE
to be used. Thus it doesn't breach any RFC, if a device is using
another packetisation nevertheless.<br>
<br>
BR<br>
Jan<br>
<br>
<br>
Am 08.10.2010 12:48, schrieb David Ponzone:
<blockquote cite="mid:3E86F0B9-A2F1-4F7A-979D-2C11F84EC1D7@ipeva.fr"
type="cite">
<meta http-equiv="Context-Type" content="text/html; charset=UTF-8">
Jan, no ptime means 20ms.
<div>That's the RFC.</div>
<div><br>
<div> <span>
<div><span>
<div>
<div> David Ponzone <span>Direction Technique</span>
</div>
<div> <span>email: <a moz-do-not-send="true"
href="mailto:david.ponzone@ipeva.fr">david.ponzone@ipeva.fr</a></span>
</div>
<div> <span>tel: 01 74 03 18 97</span> </div>
<div> <span>gsm: 06 66 98 76 34</span> </div>
<div> <br>
</div>
<div> Service Client<span> </span> IP eva </div>
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<div> <span>
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</div>
<br>
<div>
<div>Le 08/10/2010 à 12:40, Jan Riedinger a écrit :</div>
<br>
<blockquote type="cite">
<div> I'm terminating various destination by various
carriers. After migrating one customer to Freeswitch, we
observed problems for the termination of a specific route
for a specific carrier. I tried to examine the problem in
detail and I think it's related to problems regarding the
ptime negotiation. I think Freeswitch doesn't breach any
RFC, but I'm not sure, if the behaviour is optimal.<br>
<br>
The SDP of the Caller INVITE-Message at time 1160,056 in
the attached trace doesn't include any ptime setting.
Nevertheless Freeswitch includes a ptime=20 media
attribute in the forwarded INVITE message at time
1160,065. The ringing SDP sent by the callee at time
1161,948 again doesn't include any ptime setting.
Nevertheless, Freeswitch includes in the Session Progress
SDP (at time 1164,240) a ptime=20 media atrribute. Why try
Freeswitch to force the usage of ptime=20 for the
communication?<br>
<br>
The OK SDP at time 1164,240 again doesn't contain a ptime
media attribute. Nevertheless, the Freeswitch add a
ptime=20 media attribute forwarded to the caller at
1164,256.<br>
<br>
It seems that the callee is sending in the following with
a frame size of 60 bytes - it never claimed to use
ptime=20 and according the RFC 3264 it SHOULD send with
ptime=20 because of the received INVITE message
specification, but it DON'T HAVE to send with ptime=20. <br>
<br>
At next Freeswitch tries to fix "the issue". In the
logfile I found:<br>
<blockquote>e686b430-5d2d-488b-8b58-0fca1965eea7
2010-10-07 15:20:25.673206 [WARNING] mod_sofia.c:1033 We
were told to use ptime 20 but what they meant to say was
60<br>
This issue has so far been identified to happen on the
following broken platforms/devices:<br>
Linksys/Sipura aka Cisco<br>
ShoreTel<br>
Sonus/L3<br>
We will try to fix it but some of the devices on this
list are so broken,<br>
who knows what will happen..<br>
</blockquote>
This log message isn't correct. The callee never specified
anything about the usage of a specific ptime. Furthermore,
according RFC 3264 the ptime doesn't specify the frame
size, which will be used to send packages by the side,
which specify it in the SDP. In the RFC 3264 is written:<br>
<pre> If the ptime attribute is present for a stream, it indicates the
desired packetization interval that the offerer would like to
receive .
...
There is now requirement that the packetization interval be the same in each direction for a particular stream.</pre>
<br>
IMHO that means, that it isn't possible in principle that
a device is lying about it's ptime usage, because it only
specify by the media attribute the packetization it likes
to receive and doesn't specify the packetization it will
use itself. <br>
<br>
For fixing "the problem" Freeswitch sends a re-INVITE
message at 1164,777. This message includes in the message
header "X-Broken-PTIME: Adv=20; Sent=60", and ptime = 60
media attribute.<br>
The callee fails to process this re-INVITE and drops the
call.<br>
<br>
I made the trace after I set the newly introduced
parameter "passthru_ptime_mismatch=true" (it's documented
in the Wiki since yesterday). Does it make sense, that
Freeswitch tries to fix any ptime setting if this variable
is set to true?<br>
<br>
If someone wants to examine this issue more detailed, I
can provide the Wireshark-cap file of the call and the
debug output of Freeswitch.<br>
<br>
<br>
Thank you in advance<br>
Jan<br>
<br>
<br>
<br>
<br>
<br>
<pre>--
Jan Riedinger Phone : +49-30-39 73 19 66
Dipl.-Inf. | Managing Director Fax : +49-30-39 73 19 64
E-Mail: <a moz-do-not-send="true" href="mailto:riedinger@sns.eu">riedinger@sns.eu</a>
SNS Consult GmbH ICQ : 163-237-041
Südwestkorso 49a MSN : <a moz-do-not-send="true" href="mailto:jan@sns-consult.de">jan@sns-consult.de</a>
14197 Berlin GERMANY Skype : Jan Riedinger
AG Charlottenburg - HRB 71973
</pre>
</div>
<span><Ptime Problem Call Trace.tif></span>_______________________________________________<br>
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</blockquote>
</div>
<br>
</div>
<pre wrap="">
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</pre>
</blockquote>
<br>
<pre class="moz-signature" cols="72">--
Jan Riedinger Phone : +49-30-39 73 19 66
Dipl.-Inf. | Managing Director Fax : +49-30-39 73 19 64
E-Mail: <a class="moz-txt-link-abbreviated" href="mailto:riedinger@sns.eu">riedinger@sns.eu</a>
SNS Consult GmbH ICQ : 163-237-041
Südwestkorso 49a MSN : <a class="moz-txt-link-abbreviated" href="mailto:jan@sns-consult.de">jan@sns-consult.de</a>
14197 Berlin GERMANY Skype : Jan Riedinger
AG Charlottenburg - HRB 71973
</pre>
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