[asterisk-users] How to enter bridge_native_loop???

[Steve Totaro]

Olle E Johansson wrote:
>
> 9 mar 2007 kl. 21.14 skrev Santosh Raghuram:
>
>> Hi,
>>
>> With canreinvite=yes, all the media/rtp traffic for the call 
>> typically flows directly between the two peers. So how is the code in 
>> bridge_native_loop called and when? Is it called and used for any 
>> further sip signalling and not rtp?
>>
>>
> We have a couple of RTP bridges. SIP is not bridged. The RTP bridge is 
> normally used only
> for calls between two channels that use RTP, like SIP, Gtalk, H.323 
> and MGCP - depending on
> the channel code. The Core bridge is used whenever we have 
> incompatible channels.
>
> As you say, the default behaviour for the SIP channel is to make sure 
> media is passed outside
> of the Asterisk box. There's a few things that stop this behaviour, 
> making Asterisk keep the
> media flowing through Asterisk.
>
> 1) A need to listen to DTMF
> 2) Something else that needs media (monitoring the call)
> 3) NAT support or canreinvite=no
>
> If we determine that the only reason to keep the media inside Asterisk 
> is NAT support
> or that external media is disabled by canreinvite=no, then we try the 
> p2p RTP bridge
> where Asterisk basically is an RTP forwarder, very much like Sip 
> Express Router's and
> OpenSER's RTP proxies.
>
> When SIP set's up the call, it calls the RTP bridge in rtp.c that 
> determines whether we
> can do a RTP 2 RTP bridge or let the core bridge take over. Secondary, 
> it checks
> whether any of the call legs has one of the above issues. If not, the 
> remote bridiging
> (external media) kicks in and SIP sends out re-invites. If that 
> doesn't work (see
> list above), the bridge checks case 1 and 2 - if they do not apply, we 
> use p2p RTP
> bridging.
>
> I am a bit unsure on what happens if you force a jitterbuffer in this 
> case. I would
> assume that the p2p RTP bridge is turned off in that case.
>
> Remember that each channel/media driver has different ways to handle 
> this.
> Zapata has native bridging too, but different rules apply depending on 
> the
> technology used.
>
> /Olle
>
>
> ---
> * Olle E. Johansson - oej at edvina.net
> * Asterisk Training http://edvina.net/training/
> * Asterisk SIP Masterclass, Stockholm May 7-11 - register now!
Wow Olle!  You should teach a class!  Oh wait, you do!