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<div class="moz-cite-prefix">On 03/24/2015 04:28 PM, Richard Mudgett
wrote:<br>
</div>
<blockquote
cite="mid:CALD46g3aJFmfibOqjA2AgtkhBAmaTnCogCRPjY8pQONu_f03cw@mail.gmail.com"
type="cite">
<div dir="ltr"><br>
<div class="gmail_extra"><br>
<div class="gmail_quote">On Tue, Mar 24, 2015 at 4:17 PM, Jeff
LaCoursiere <span dir="ltr"><<a moz-do-not-send="true"
href="mailto:jeff@jeff.net" target="_blank">jeff@jeff.net</a>></span>
wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0
.8ex;border-left:1px #ccc solid;padding-left:1ex"><br>
Hello,<br>
<br>
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.<br>
</blockquote>
<div><br>
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<div>That will only happen when the call is natively
bridged:<br>
<br>
</div>
<div>Non-native bridge: Packets can get translated or
Asterisk has an interest in the packet for things like
DTMF or call recording.<br>
</div>
<div>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.<br>
</div>
<div>Native bridge doing direct media (Remote bridging):
Packets go directly between endpoints so Asterisk never
sees them.<br>
<br>
</div>
<div>Richard<br>
</div>
</div>
<br>
</div>
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</blockquote>
<br>
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.<br>
<br>
Cheers,<br>
<br>
j<br>
<br>
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