[asterisk-users] Advanced Codec Negotiation: Need info and uses cases

[Jon Bonilla (Manwe)]

El Tue, 9 Jun 2020 09:46:32 -0600
George Joseph <gjoseph at digium.com> escribió:


Hi George

> 
> 
> >
> > If transcoding is enabled Would it be possible to do the same but handle a
> > 488
> > back from Bob and failover to another INVITE with Bob's allow list to
> > handle
> > transcoding? That way we would always try no-transcoding before offering a
> > transcoding codec.
> >  
> 
> I think this is a good idea but might not be possible in the first release
> of Advanced Codecs.
> We'd also probably need an option to enable/disable retrying.
> 
> We might also be able to just handle this by taking Bob's allowed codecs
> but sorting
> them so that any codecs that Bob and Alice share come at the front of the
> list
> in the offer to Bob.
> 
> 

In this scenario. For example if we have

[alice]
disallow=all
allow=alaw
allow=opus

[bob]
disallow=all
allow=opus
allow=alaw
allow=gsm


In case Alice sends an INVITE with opus and g723 in the sdp.

1: We remove g723 as it's not in Alice's allow

2: When we send the INVITE to Bob either we:

  2a: Send the invite only with opus to avoid transcoding. If Bob's phone
  doesn't support opus and sends 488 back I'd send another INVITE with alaw and
  gsm to handle the call with transcoding.

  2b: We send the invite with opus, alaw and gsm and let Bob choose. In this
  case we don't try to avoid transcoding. 

  2c: We send invite with opus only and don't retry. I would do this only if
  transcoding is disabled by configuration. 

My preferred solution would be 2a/c based on a transcoding on/off configuration
option.

In your explanation transcoding is not done if you filter by the allowed codecs
of both + alice's INVITE. It could be possible to receive a 488 and without
retry we lose any transcoding capability.


right?


cheers,

Jon


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