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We seem to have a problem with Asterisk 1.4 when a client sends through
their SDP information but includes encoding parameters on the end of
their SDP information. For example some phones send:<br>
<br>
a=rtpmap:18 G729/8000/1<br>
<br>
instead of the usual:<br>
<br>
a=rtpmap:18 G729/8000<br>
<br>
in the SDP...<br>
<br>
It seems that when the encoding parameter '/1' is included at the end
of the rtpmap statement, Asterisk doesn't recognise the codec
internally and then has trouble transcoding giving errors such as
'Unable to find a codec translation path from unknown to unknown'.
'sip show channels' also shows the 'Form' as 'unkn' during a call.
This behaviour only appears to happen though when the encoding
parameter is included. According to RFC2327:<br>
<tt><br>
</tt>
<blockquote><tt>The general form of an rtpmap attribute is:</tt><br>
<br>
<tt>a=rtpmap:<payload type> <encoding name>/<clock
rate>[/<encoding parameters>]</tt><br>
<br>
<tt>For audio streams, <encoding parameters> may specify the
number of</tt><br>
<tt>audio channels. This parameter may be omitted if the number of</tt><br>
<tt>channels is one provided no additional parameters are needed. For</tt><br>
<tt>video streams, no encoding parameters are currently specified.</tt><br>
</blockquote>
<br>
So, the encoding parameter part looks like an optional but perfectly
valid part of the rtpmap SDP definition.<br>
<br>
Interestingly calls often seem to work fine out to the PSTN etc. but
Asterisk has problems transcoding between 2 local clients. Has anybody
seen this behaviour in Asterisk 1.4? Is this a bug or a feature that I
haven't setup in Asterisk I am yet to discover?<br>
<br>
Cheers,<br>
Ray<br>
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