I have captured the logs at both ends for a end-to-end Wi-Fi call using two phones.
They have mainly SIP and RTP packets. I am getting jitter between 5-15 ms for reverse and forward directions. I have a local router which isn't connected to LAN.
Any help is greatly appreciated. Thanks!
asked 11 Jul '12, 12:16
This will show all RTP streams and Min/Max Jitter for each stream (scroll to the right).
Then select one stream and click on
Well, it depends ... Some network providers offer SLAs with 0.5 - 2 ms max jitter. That's pretty good. Others (e.g. Avaja, Cisco) say, 10-20 ms is acceptable. Furthermore there are compensations techniques, like jitter buffers. So a jitter value of 15 ms can cause problems (crippled audio) in one environment and no problems at all in another environment (with jitter buffers).
As I mentioned, Jitter can be caused by numerous factors. Even the VoIP devices (especially soft phones) can cause jitter due to process scheduling in the device. So, if there are no signs of other network problems, you should consider the VoIP devices as a possible source.
As you mentioned that the VoIP endpoints are connected via Wifi, I suggest to check that connection first. A Wifi link can also cause jitter, especially if the network is "crowded" or if there are other interfering radio signals. You can test the jitter of the network with xjperf (UDP tests). I'm not sure how iperf calculates the jitter value, so it may not be comparable with the VoIP jitter value! However, it's something to start with.