[Asterisk-Users] Interop RTP "Extension headers" for QOS?

[Matt Schulte]

Yah I have wondered that also. All of our devices do it, PSTN --> SIP,
PSTN --> MGCP and vice versa. How long have you been stuck on the
problem? And yes, we're dealing with the "test case" as well. :-) We had
to turn off reinvites, the team at Interop mentioned someone else was
having problems with reinvites. He was confused why we were sending so
many invites *shrug*. I think he's new to the field, then again aren't
we all? 

	Matt

-----Original Message-----
From: Ken Wiesner [mailto:Ken.Wiesner at logicalonline.com] 
Sent: Friday, August 13, 2004 8:33 AM
To: Matt Schulte
Subject: RE: [Asterisk-Users] Interop RTP "Extension headers" for QOS?


Matt,

I have this same problem too and they told me the same thing.  However,
I'm wondering if it has something to do with Asterisk performing a
native bridge of the RTP traffic.  I'm assuming you were also handed the
60 question form?  I've been having a hard time getting through the
testing.  I can't seem to get re-invites to work properly.  Have you had
any luck with this?

~ken

-----Original Message-----
From: asterisk-users-admin at lists.digium.com
[mailto:asterisk-users-admin at lists.digium.com] On Behalf Of Matt Schulte
Sent: Friday, August 13, 2004 6:58 AM
To: asterisk-users at lists.digium.com
Subject: [Asterisk-Users] Interop RTP "Extension headers" for QOS?

We're setting up a connection with Level3's voip system and when we use
Asterisk or make or recv calls we get an initial pulsing noise. Level3's
Interop team explains that's their RTP extension headers and Asterisk
apparently doesn't know what to do with it. He said we need to either
ignore or of course let the traffic pass. Has anyone heard of this
before? I understand the concept but have no clue where to start
looking. I've been googling all last night about it and have come up
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