I am trying to set-up an Asterisk server to "transcode" between different RTP
frame sizes. I have devices on one side using alaw with ptime 20 ms and some
equipment on the other side requiring ptime 10. I am using the latest Asterisk
13.8.0. The set-up looks like this:
A (ptime 20) ---> asterisk ---> B (ptime 10)
In Asterisk I have two peers defined with only one codec in each, alaw:10 and
alaw:20 respectively. The SIP call set-up looks fine and each side announces
the correct ptime in the SDP (both Asterisk and B has the ptime=10 attribute in
the SDP). The dialplan is currently first answering A's call, plays a prompt
and then Dial() into B.
This is what the media streams look like, including RTP frame size:
A --- 20ms -------> asterisk -----20ms!-----> B
A <-- 20ms ------- asterisk <-----10ms---- B
The stream from Asterisk to B has the wrong frame size, it should be 10ms.
Looking at the media from B to A, we can see that asterisk properly changes
frame size in one direction.
I also tried to use ulaw on the path between Asterisk and B to see if that
would trigger a proper transcoding, but the results were the same (in terms of
frame size, but with correct change of codec ).
Is this supposed to work? Any suggestions for workarounds?
Best regards,
Jan Blom
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