AJAM & Call Quality

Asterisk Asterisk Support
1

Hi

I was using AJAM and realized that during each call, a constant flow of events “RTCPReceived” was sent, holding info about call quality (jitter, lost data, rtt…).

Also, at the end of each call, two events are fired with global stats about the call. (examples below).

  • Where can I find a description of the fields ?
  • Can the Stat data be written to a file via simple config (I’m considering writng a small prog to do this, but I hate reinventing the wheel)
  • Does anyone know a program that would make use of this to generate call quality stats ?

thanks for the feedback,

J.

dlsr: "369.8600(sec)" event: "RTCPReceived" fractionlost: "0" from: "xxx:19489" highestsequence: "49977" iajitter: "72" lastsr: "7160.125965972422000640" length: 0 packetslost: "87246" privilege: "reporting,all" pt: "200(Sender Report)" receptionreports: "1" rtt: "102(sec)" senderssrc: "3461611776" sequencenumbercycles: "0"

event: "RTPReceiverStat" jitter: "0.0019" length: 0 lostpackets: "19851" privilege: "reporting,all" receivedpackets: "800" rrcount: "101" ssrc: "1982888845" transit: "0.0161"

2

Howdy,

  1. The only description may be in the source itself.
  2. No, the data is only output over AMI right now. If you want to write it to file, you’ll have to capture it yourself and write it.
  3. You may not want to take the data too seriously; right now it does not generate everything correctly. There was a community project underway to improve this, but I have not seen progress on it in quite some time.

Cheers.

3

I had a feeling that the anwers for 1 & 2 would be this… 3. is a bit disappointing… tough life !

many thanks for this valuable info,

J

4

Howdy,

Check into this branch:
svnview.digium.com/svn/asterisk/ … rog-trunk/

It was the improvements work, but it’s a bit dated at this point.