Hi folks<br><br>I have a handset talking to Asterisk, which in turn puts the call through to an ITSP.<br><br>The handsets likes to send audio in 40ms frames (even though Asterisk requests 20ms frames with a ptime header in the SDP).<br>
<br>The ITSP doesn't request any particular frame length (with ptime) or state a maximum length (with maxptime), so when Asterisk receives the 40ms media frames from the handset, it simply relays it on to the ITSP. Unfortunately the ITSP doesn't support this, and the result is one-way audio.<br>
<br>I would like to know whether there is a way to force Asterisk to repacketize the media stream, converting from 40ms frames to 20ms frames. I am aware of the allow=alaw:20 syntax but this doesn't seem to work. It is not clear from the docs whether that setting is for the SDP offer (in which case it would only affect incoming media) or whether it's used to to actually force what is sent to a peer/user as well (in which case it is not behaving as expected).<br>
<br>I would also be interested to know what the correct action is for the ITSP to take. How do they complain (using SIP and/or SDP) that the media received is not what they were expecting? <br><br>Any help would be greatly appreciated.<br>
<br>Regards,<br>Richard<br><br clear="all">--<br>Richard Brady<br>T: +44 (0)7771 623 348<br>E: <a href="mailto:rnbrady@gmail.com">rnbrady@gmail.com</a><br>