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 - [EMAIL PROTECTED]
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