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<p class=MsoNormal>All,<o:p></o:p></p>
<p class=MsoNormal><o:p> </o:p></p>
<p class=MsoNormal>We have a vendor which insists that the standards don’t
forbid this, so they won’t change the behavior, but it simply defies the
intent of the RFC in my opinion.<o:p></o:p></p>
<p class=MsoNormal><o:p> </o:p></p>
<p class=MsoNormal>SIP dialogs set up two RTP streams between (for all intents)
two endpoints and a media server (which mixes the audio and sends to a “conference-in”
recording server, but that’s not relevant unless we dig into the RTP
standard).<o:p></o:p></p>
<p class=MsoNormal><o:p> </o:p></p>
<p class=MsoNormal>Early in the call, in order to play a tone/message to one of
the endpoints, the media server plays the tone Endpoint A while simultaneously
putting RTP Stream B (to Endpoint B) in a loopback state (by changing the
destination IP of the stream to itself) so no audio is played to Endpoint B,
and after 1s or so re-point RTP stream to Endpoint B (by changing back the
destination IP). The effect is this: Monitoring and reporting tools see the
initial RTP packets with consecutive sequence numbers, which disappear for ~1s,
then reappear with incremented sequence numbers… thus appearing to the
monitoring/reporting tools as 1s of packet loss.<o:p></o:p></p>
<p class=MsoNormal><o:p> </o:p></p>
<p class=MsoNormal>I’m having difficulty providing solid RFC statements
which tell this vendor that they’re doing something (changing the
destination IP address during the call) which violates the entire point of the
RTP sequence number (which is to identify packet loss), and then vendor is
being obtuse.<o:p></o:p></p>
<p class=MsoNormal><o:p> </o:p></p>
<p class=MsoNormal>Can anyone provide an argument which can be used here?<o:p></o:p></p>
<p class=MsoNormal><o:p> </o:p></p>
<p class=MsoNormal>Nathan<o:p></o:p></p>
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