asterisk users - Apr 2016 - Changing RTP frame size

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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On Thu, Apr 7, 2016 at 11:04 AM, Jan Blom <jan.blom at
peopleinteractive.se> wrote:
> <snip>
> Is this supposed to work? Any suggestions for workarounds?
I believe so. That sounds odd. Hard to know without seeing the packet
trace of the call.

Which SIP channel driver are you using?

I think you are safe to go ahead and file an issue report. Please
include the sip.conf/pjsip.conf plus a packet capture and Asterisk
debug log (be sure to get the DEBUG channel turned on in logger.conf)
with correlating SIP trace.

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Rusty Newton
Digium, Inc. | Community Support Manager
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