[asterisk-dev] Testers wanted (I mean YOU) for Comfort Noise labs

[Olle E. Johansson]

19 nov 2012 kl. 18:03 skrev Matthew Jordan <mjordan at digium.com>:

> On 11/19/2012 07:17 AM, Olle E. Johansson wrote:
>> Friends,
>> 
>> My work with comfort noise support in Asterisk took a big leap forward today. I had a version that
>> replaced the lack of frames with music on hold, but today it was swapped out and I now use the
>> noise generator contributed by cmantunes earlier. 
>> 
>> This only applies to the core bridge (at least that's what I tested). I will investigate what's needed
>> for app_queue and possibly if needed, the rtp bridge.
>> 
>> Asterisk will properly negotiate CN support in SDP and will take incoming CNG frames and
>> either forward over the bridge (if the other side of the call supports CN) or generate noise
>> until audio frames are received.
>> 
>> I have not added a silence generator, so asterisk will not generate it's own CN frames and
>> suppress silence. This is a coming step. At least we can now bridge a call with full CN 
>> support.
>> 
>> If you want to help me test, check out this branch
>> http://svn.digium.com/svn/asterisk/team/oej/roibos-cng-support-1.8
>> 
>> Check the README.roibos-cng.txt file for more information.
>> 
>> A big thank you to Matt Jordan and Joshua Colp for guidelines and ideas on this work.
>> 
> 
> Cool stuff Olle.
> 
> Are there some particular scenarios and/or endpoints you'd like to see
> tested?
All endpoints. All dialplans that I can't imagine.

For scenarios - I've been focusing on the core bridge. We do have other bridges
and I would like to see if this works or not. Next step for me is to dig into
the decision about core/native RTP bridge and determine that if both
devices in a bridge have CN support, allow native, if not, keep it in core
bridge (since transcoding may be and propably will be needed from
SLINEAR).

Question: If I run RTP bridge (not the p2p or remote) can we still operate
on timer? If not, I have to disable RTP bridging totally. If we rely on incoming
packets (which will not happen) to send out, CN will not work.

> 
> Are there any scenarios that you think could be automated with SIPp?
> (If nothing else, it looks like the SDP negotiation could be put into a
> SIPp test pretty quickly)
We would need to add scenarios for the SDP negotiation:
- A with no CN calls B with CN
- A with CN calls B with CN

We need to set up calls with a dial option that excludes a native bridge
and allowing a native bridge.

One test is of course setting up the SDP agreement. The second is to
check the RTP stream and see what happens.

> 
> Currently the Asterisk Test Suite assumes a team branch is equivalent to
> trunk, but we could tweak it to allow it to 'force' the Asterisk
> installation under test to be treated as a particular version.  That
> would allow a Bamboo plan to be pointed at your branch and have the 1.8
> compatible tests run against it.  Any scenarios that are automated could
> be dropped into a branch of the Test Suite and run against the roibus
> Asterisk branch as well.
> 
> Let me know if you're interested and I'll get that set up.
THat would be very cool. Thank you.

Note to the rest of you: Matt's help doesn't take you guys and dolls of the hook. Please test :-)

/O