[asterisk-users] Setting up RTP to flow between endpoints directlybypassing Asterisk

[Klaus Darilion]

Am 16.03.2010 01:42, schrieb Jeff Brower:
> Vikram-
>
>> http://www.voip-info.org/wiki/view/Asterisk+Letting+SIP+clients+connect+directly
>>
>> The link above indicates that it is possible to setup RTP streams to
>> directly flow between endpoints and completely bypass Asterisk. I would
>> like to know if this configuration would work when,
>>
>> a) both endpoints are behind NAT, and/or
>> b) both endpoints don't support same codecs
>>
>> with media flowing through a SIP+rtpproxy server that can do
>> transcoding ?
>
> This would be 'native bridging' mode as I've seen it described a few places on the web, correct?  If Asterisk is "out
> of the RTP loop", then what can it still do?  Only billing?  It would not detect DTMF, no RTP record or announcement
> playout, etc.

No, this this is not native bridging.

Asterisk supports 3 methods of media handling:
1. bridging: media (audio, video) is received on one channel, handled 
over to Asterisk's core, forwarded to the bridged channel, and sent out 
again.

2. native-bridging: if both bridged channels use the same technology 
then media can be bridged directly in the channel driver, no need to 
feed the media into Asterisk's core. For example SIP-to-SIP calls or 
DAHDI-to-DAHDI calls.

3. bypass: here, the media flow bypasses Asterisk directly. AFAIK this 
works only with SIP as channel technology. This comes in 2 flavors:

  3a) During call setup the media will be forwarded via Asterisk. Once
the call is set-up, Asterisk will send reINVITEs to both clients using 
the clients original SDP contact information. For this you must set 
canreinvite=yes (1.4) or directmedia=yes (1.6) in sip.conf. Of course 
Asterisk will initiate the direct media only if the media is not needed 
in Asterisk, e.g. if you monitor a call, the media will always be routed 
via Asterisk.

  3b) Media will bypass Asterisk from the beginning. Therefore you have 
to set directrtpsetup=yes. This is still experimental and causes weird 
reINVITEs (e.g. after call setup to lock down on a certain codec or 
after call termination to redirect media to Asterisk before hanging up).

Both bypass modes Note only work if either there are no NATs at all, or 
the clients are behind the same NAT and do not use STUN.

regards
Klaus