[asterisk-dev] [Code Review] Improved support for T38 on initial INVITE

Mark Michelson mmichelson at digium.com
Wed Mar 25 14:35:02 CDT 2009


-----------------------------------------------------------
This is an automatically generated e-mail. To reply, visit:
http://reviewboard.digium.com/r/208/#review626
-----------------------------------------------------------

Ship it!


Looks good to my eyes. The positive testing improves my confidence, too. Good job, Josh!

- Mark


On 2009-03-25 09:27:37, Joshua Colp wrote:
> 
> -----------------------------------------------------------
> This is an automatically generated e-mail. To reply, visit:
> http://reviewboard.digium.com/r/208/
> -----------------------------------------------------------
> 
> (Updated 2009-03-25 09:27:37)
> 
> 
> Review request for Asterisk Developers.
> 
> 
> Summary
> -------
> 
> This change improves our handling of T38 when we get it on an initial INVITE.
> 
> Previously we would try to setup a T38 only session end to end (between both endpoints) if the incoming call offered T38 (even if it also offered audio).
> 
> This change makes chan_sip respond appropriately to the incoming INVITE. If we get an offer with both T38 and audio, we setup both a T38 and audio session.
> We do not, however, setup a T38 and audio session on a subsequent outgoing channel. We setup an audio only session. If the incoming channel sends us UDPTL
> though we immediately trigger a reinvite on the outgoing channel to T38.
> 
> This seems to be the best compromise I can come to.
> 
> 
> Diffs
> -----
> 
>   /branches/1.4/channels/chan_sip.c 184187 
> 
> Diff: http://reviewboard.digium.com/r/208/diff
> 
> 
> Testing
> -------
> 
> A few users have tested this change in their environments to confirm it works. I have also confirmed that the new behavior is present when T38 is present in the initial INVITE with my T38 test setup here.
> 
> 
> Thanks,
> 
> Joshua
> 
>




More information about the asterisk-dev mailing list