No subject

Our Receiver
	ssrc          our ssrc
	rxcount       no. received packets/Received packets
	lp            lost packets/Lost packets
	rxjitter      our calculated jitter(rx)/Jitter
=09
Our Sender
	themssrc      their ssrc
	txcount       transmitted packets/Sent packet
	rlp           remote lost packets/Lost packets
	txjitter      reported jitter of the other end/Jitter
	rtt           round trip time/RTT
=09

   Synchronization source (SSRC): The source of a stream of RTP
      packets, identified by a 32-bit numeric SSRC identifier carried in
      the RTP header so as not to be dependent upon the network address.
      All packets from a synchronization source form part of the same
      timing and sequence number space, so a receiver groups packets by
      synchronization source for playback.  Examples of synchronization
      sources include the sender of a stream of packets derived from a
      signal source such as a microphone or a camera, or an RTP mixer
      (see below).  A synchronization source may change its data format,
      e.g., audio encoding, over time.  The SSRC identifier is a
      randomly chosen value meant to be globally unique within a
      particular RTP session (see Section 8).  A participant need not
      use the same SSRC identifier for all the RTP sessions in a
      multimedia session; the binding of the SSRC identifiers is
      provided through RTCP (see Section 6.5.1).  If a participant
      generates multiple streams in one RTP session, for example from
      separate video cameras, each MUST be identified as a different
      SSRC.

Regards,
Mindaugas Kezys
http://www.kolmisoft.com
VoIP Billing and Routing Solutions


-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of hh174
Sent: 2009 m. rugs=EBjo 5 d. 17:37
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] ${CHANNEL(rtpqos,audio,all)}

Hi all,
With Asterisk 1.6.1.6
Trying to have statistic concerning Rtp audio quality, I use=20
${CHANNEL(rtpqos,audio,all)}
having also tried AUDIORTPQOS and ${CHANNEL(rtpqos,audio,...)}


Sometimes, it works and I have results.
Most of the time I get strange or no results even when the call was=20
succesfull.
rtpdest set at 0.0.0.0:0, no Joitter information, no packetlosts,...

It seems that when the channel is hungup, some informations are lost=20
(often the cas with rtpdest) depending on the party hanging-up.

Also, some info are not clear for me, like what are the meaning of
-rtt? (Delay?)
-ssrc=3D1271016709 (what is the meaning of this number?
-themssrc

Any clue, docs, informations to make the rtp statistics working?
What do I wrong?

Olivier




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