[asterisk-dev] [Code Review] more reliable sip STUN support

[Klaus Darilion]

Am 04.08.2010 19:52, schrieb David Vossel:
> ----- Original Message -----
>> From: "Kevin P. Fleming"<kpfleming at digium.com>
>> To: asterisk-dev at lists.digium.com
>> Sent: Wednesday, August 4, 2010 12:04:36 PM
>> Subject: Re: [asterisk-dev] [Code Review] more reliable sip STUN support
>> So it sounds like we should step back and try to document exactly what
>> Asterisk is using the STUN client mechanism to accomplish, because
>> there
>> may be some confusion about that.
>
> Agreed, I'll figure this out and start a new discussion once I have it documented.

Maybe you can extend chan_sip to use STUN for the RTP sockets too

regards
Klaus