IAX2 support

[haralambop]

No description provided.

[meetecho]

I'm unfamiliar with IAX2, so I'm not sure how codecs are negotiated there. Something that needs to be considered is that Opus always negotiates a 48kHz rate, even when it's going to work at lower (e.g., narrow or wideband) ones. Do you know if this might be a dealbreaker for IAX2? When you mean "no voice coming on the other end", do you mean only silence comes out or no RTP is flowing at all?

[meetecho]

I just commited a fix that may be related, as it caused issues with RTP and audio in general: could you try with the new patch?

[haralambop]

Voice is now ok but the bandwidth needed is much higher then g729.

Besides i receive following error

[Opus] Created encoder #7 (48000->opus)
[Opus] Created decoder #7 (opus->48000)

000003]: translate.c:362 framein: opustolin48 did not update samples 0
[Opus] Ops! got an error decoding

[Opus] Destroyed encoder #6 (48000->opus)
[Opus] Destroyed decoder #6 (opus->48000)

Are you sure that the codec is working in 8000

Thank you

[meetecho]

The codec is able to seamlessly transport different rates (from 8000 to 48000) in a transparent way: in RTP, though, the codec is always reported as a 48kHz one, and Asterisk handles it accordingly. The codec integration in Asterisk takes care of "capping" the rate according to the peer: that is, if you're talking to a GSM peer, there's no sense in using a higher rate, and so Opus encoding is capped at 8kHz as well.

In the debug you pasted, though, I see that encoders and decoders are created with a 48000, which means the codec thinks it is going to talk with an actual 48kHz peer, so this might explain why you're getting a higher bandwidth than necessary. It's likely that the flow actually is 48kHz. For the scenario I mentioned above (talking with a GSM peer) you would see a 8000->opus and opus->8000 debug.

As I said, I'm unfamiliar with IAX2 so I don't know if that's the cause of the issue, that is, the protocol making Asterisk believe it's actually 48kHz. If you check the asterisk-dev mailing list, and in particular this post from a longer thread discussing this work:

http://lists.digium.com/pipermail/asterisk-dev/2013-June/060591.html

you'll see that other users are able to get this to work: nevertheless, I don't think they're using Opus as the codec for trunking (as I think is your case).