T.140 issues w/Asterisk 13

Asterisk Asterisk SIP
1

Attempt to use T.140 (and T.140red) in Asterisk 13. Whenever Asterisk sends a T.140 packet it seems to append “RTP” on a newline to every message.

Also, noticing when it sends an OK back from an invite it lists T.140 above RED, but sends RED packets anyway.

Any suggestions on either?

2

To explain better the message above:

  1. In Asterisk 13 and 16 RTT, whenever Asterisk sends T.140 message the string “RTP” and null terminator are written to the end of the message. This also affects the packet data in T.140 RED. Is this defect in Asterisk code because the spec and text capabable clients such as SipCon1 do not append the string “RTP” after every character or set of characters is sent?
  2. Another issue, If just a T.140 invite comes into Asterisk, Asterisk will connect the call initially, but disconnects with the message “Bearer Capability Not Available” and the Asterisk CLI says codec not available. Can Asterisk T.140 call with no audio in the SDP?
3

T.140 is rarely used, so it’s entirely possible that the functionality has issues. I do recall someone putting some time into it fairly recently, but don’t remember further details.

Asterisk is very audio heavy and really requires an audio stream to be present.

4

Does anyone have a recommend way to debug Asterisk? I am not a Linux developer, but have C/C++ knowledge. The frame.c has a function that duplicates the frame and it adds the frame.src right after the frame data which is thus RTP\0. However, that would mean that the length of the data was incremented by four somewhere after that function is called or something else is causing it.

5

That would be better on the developer mailing list or IRC channel.

Most of my debugging was done by code reading, backed up by the occasional single stepping with gdb.

6

This topic was automatically closed 30 days after the last reply. New replies are no longer allowed.