[asterisk-users] Forcing repacketization on SIP to SIP call

[Richard Brady]

Hi folks

I have a handset talking to Asterisk, which in turn puts the call through to
an ITSP.

The handsets likes to send audio in 40ms frames (even though Asterisk
requests 20ms frames with a ptime header in the SDP).

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.

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).

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?

Any help would be greatly appreciated.

Regards,
Richard

--
Richard Brady
T: +44 (0)7771 623 348
E: rnbrady at gmail.com
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