[asterisk-users] RTP handling
rmudgett at digium.com
Tue Mar 24 17:07:12 CDT 2015
On Tue, Mar 24, 2015 at 4:59 PM, Jeff LaCoursiere <jeff at jeff.net> wrote:
> 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> wrote:
>> 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.
> 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.
You cannot force native bridging. It will switch to native bridging if you
don't set anything
that makes Asterisk interested in the media stream. Such as enabling
DTMF features in features.conf and Dial flags like t or T.
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