RE: [Iaxclient-devel] Re: Bug info... (really weird bug info)

[Steven S. <ss...@so...>]

FYI - I tried setting notransfer=yes and it had no effect.  Asterisk still
tried to establish a native bridge.  Still lost audio after 65 seconds.

> -----Original Message-----
> From: iax...@li... [mailto:iaxclient-devel-
> ad...@li...] On Behalf Of Steven Sokol
> Sent: Sunday, February 08, 2004 2:43 PM
> To: iax...@li...
> Subject: [Iaxclient-devel] Re: Bug info... (really weird bug info)
> 
> More Data:
> 
> Watching the trace, the error seems to take place after the timeout for
> the
> transfer attempt.  As Adam pointed out, IAX2 always tries to establish a
> native bridge.  Somehow our clients, being both NATed behind the same NAT
> device (i.e. both using the same real outside IP address) are unable to
> connect natively (the NAT router is not smart enough to "pinwheel" IAX
> packets?) so the connection does not work.
> 
> Oddly, the IAX protocol seems to default to switching over to a
> native-bridged operation UNLESS it receives tacit failure messages from
> both
> sides.
> 
> Here's what you see with a IAX2-to-IAX2 call through Iaxtel that does
> _not_
> have any problems with the audio:
> 
>     -- IAX2[1111]/8 answered IAX2[iaxtel@69.73.19.178:4569]/7
>     -- Stopped music on hold on IAX2[iaxtel@69.73.19.178:4569]/7
>     -- Attempting native bridge of IAX2[iaxtel@69.73.19.178:4569]/7 and
>     -- Channel 'IAX2[iaxtel@69.73.19.178:4569]/7' unable to transfer
> 
> Note the last line - the chan_iax2 code at Iaxtel detected that a native
> session could not be established, so it rejected the attempt and my
> Asterisk
> continued to relay the voice data.
> 
> Here's what you see with the IAX client-to-IAX client call:
> 
>     -- Accepting unauthenticated call from 64.151.42.28, requested format
> =
> 2, actual format = 2
>     -- Executing Dial("IAX2[guest@1105]/3", "IAX2/1111|20|m") in new stack
>     -- Called 1111
>     -- Started music on hold, class 'default', on IAX2[guest@1105]/3
>     -- Call accepted by 64.151.42.28 (format GSM)
>     -- Format for call is GSM
>     -- IAX2[1111]/5 is ringing
>     -- IAX2[1111]/5 answered IAX2[guest@1105]/3
>     -- Stopped music on hold on IAX2[guest@1105]/3
>     -- Attempting native bridge of IAX2[guest@1105]/3 and IAX2[1111]/5
>     -- Registered '1111' (AUTHENTICATED) at 64.151.42.28:4569
>     -- Registered '1111' (AUTHENTICATED) at 64.151.42.28:4569
> 
> You never see either side (properly?) reject the attempted native
> transfer,
> so Asterisk bows out of the call and the audio stops.
> 
> Can we set a noreinvite flag for IAX (I know, I should look at the
> source...) to prevent the system from event trying to reinvite the audio
> point-to-point?
> 
> I want to blame the "both clients behind the same NAT screen" factor, but
> I
> have had client-to-client calls loose audio after 1 minute even when users
> are calling in from outside.
> 
> Thoughts?
> 
> Adam, did you have anything like this spring up in Firefly?
> 
> Thanks,
> 
> Steve
> 
> 
> 
> 
> 
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