[asterisk-users] RTP handling

[Jeff LaCoursiere]

On 03/24/2015 04:28 PM, Richard Mudgett wrote:
>
>
> On Tue, Mar 24, 2015 at 4:17 PM, Jeff LaCoursiere <jeff at jeff.net 
> <mailto:jeff at jeff.net>> wrote:
>
>
>     Hello,
>
>     I am wondering if asterisk does anything at all to RTP packets
>     passed from channel to channel if no transcoding is involved? Can
>     I assume that the packet that left phone A, arrived at the
>     asterisk server, was copied to phone B's channel and eventually
>     arrived at phone B had exactly (byte for byte) the same payload? 
>     Assume two SIP endpoints, no NAT involved.
>
>
> That will only happen when the call is natively bridged:
>
> Non-native bridge: Packets can get translated or Asterisk has an 
> interest in the packet for things like DTMF or call recording.
> Native bridge doing packet-to-packet (Local bridging): Packets come in 
> on one channel and go out the other channel with nothing else done to 
> them.
> Native bridge doing direct media (Remote bridging): Packets go 
> directly between endpoints so Asterisk never sees them.
>
> Richard
>

Thanks for the quick reply RIchard!  Can I force native bridging, or 
does it default to that if I don't configure direct media?  The dialplan 
will be very simple - extensions calling extensions within a context.  
No DTMF, no recording, no mixing for conference, etc.

Cheers,

j

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20150324/fbe3a969/attachment.html>