[asterisk-dev] asterisk's bridge

[serva]

    I'm sorry, I have forgot to say the version of asterisk I pay attention to is 1.2.x. Not the 1.4.x one. I know that when I use 1.4.x,in ast_rtp_bridge() function, there are two differences p2p bridge,one is ture native bridge named bridge_native_loop(),in bridge_native_loop() function we also need ast_read() and ast_write(),I think this one is similar to ast_generic_bridge().The other p2p bridge is bridge_p2p_loop(),this one look like a single thread bridge, because if two same type channnels (eg. SIP channel) come into this bridge, will set voice rtp socket's fd to -1, and set a callback function, so the bridge will block by polling fd (except pipe fd),and callback function:p2p_rtp_callback() will be invoked by ast_io_wait in do_monitor function,then they can use only socket fd to send or recieve rtp.

So, I want to ask ,why not use a only one thread to do this thing instead of do_monitor, and make this thread in the core of asterisk, and make it deal with difference kinds of channel? I think this can enhance the concurrency of asterisk more better.  

------------------				 
serva
2007-11-22

-------------------------------------------------------------
From：Russell Bryant
Date：2007-11-22 23:13:27
Subject：Re: [asterisk-dev] asterisk's bridge

truename wrote:
> Hi, I found that if I set mixmonitor or spy options, that asteriosk will
> choose ast_generic_bridge() function in channel.c , but ast_rtp_bridge()
> in rtp.c can also deal with monitor and spy because monitor or spy is
> only doing in ast_read() and ast_write() function using
> queue_frame_to_spies(). So can anyone tell me , why asterisk
> choose  ast_generic_bridge() instead of ast_rtp_bridge() ?

Asterisk must be able to intercept every single audio frame.  In the case of an
RTP bridge, the audio may not be flowing through Asterisk anymore.

-- 
Russell Bryant
Senior Software Engineer
Open Source Team Lead
Digium, Inc.


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