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speedvoip
Vetran
Joined: August 07 2008
Location: Canada
Posts: 156
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| Posted: April 17 2009 at 1:15pm | IP Logged
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Hi Randal:
With media engine v6.0.0.11, there is jittered voice quality occuring frequently with codec g.729 enabled. I have uploaded packet capture into my FTP support account for your reference.
I think it is very strange!!
Thanks,
George
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support
Administrator
Joined: January 26 2005
Location: United States
Posts: 1666
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| Posted: April 17 2009 at 2:20pm | IP Logged
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Hi George,
Thanks for posting this. I will take a look and repost with more news.
Randal
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speedvoip
Vetran
Joined: August 07 2008
Location: Canada
Posts: 156
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| Posted: April 20 2009 at 12:15pm | IP Logged
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Hi Randal:
Any progress? Do I need my help?
Thanks,
George
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support
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Location: United States
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| Posted: April 21 2009 at 11:33am | IP Logged
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Hi George,
I took your log and looked at it using wireshark. The receive G729 RTP packets do experience quite a lot of jitter and roughly 8% lost RTP packets. The max receive jitter from the network is 125Ms. The jitter from your network is not good.
Using your wireshark capture log, here is what I did:
1)
Opened your capture log (Lanscape(Apr_17_2009).cap) in wireshark.
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In the main GUI, I entered the packet filter “sip and rtp” and hit the “Apply” button. I did this because I wanted to quickly look at your SIP and RTP packets.
3)
I used wireshark’s ability to perform RTP packet analysis on your RTP packets. You do this by going to the main menu Statistics->RTP->Show all streams.
A new dialog gets displayed showing all the RTP stream activity in the log. From this dialog, you can check out each RTP stream and the characteristics of each.
We can’t do much about the lost packets but you can improve your incoming audio quality if you enable jitter compensation on your VOIP Media Engine phone lines. Jitter compensation is OFF by default so you will have to enable it if you need it. Note that jitter comp will add a certain latency to your incoming audio signal when it is enabled.
To enable jitter compensation on your phone lines, call the SetJitterTimeMs() and SetEnableJitter() API procedures anytime before using the phone lines.
Based on your logs, if you set receive jitter compensation in the range of 60Ms to 125Ms, your receive audio should clean up. Enabling receiver jitter compensation should clean up your received audio at the expense of introducing audio signal path latency to your received voice signal.
Note:
We have investigated and developed a number of new packet loss concealment and jitter compensation techniques that we are working on to add to the media engine. It is our goal to get these new DSP techniques implemented some time this year. These latest DSP techniques should help to improve call quality for all media engine customers. The difference should be quite noticeable for those struggling with large RTP packet jitter issues and lost packets. If you are open to obtaining and testing these updates when they are made available, please let me know and we will work out the details.
Other issue:
When reviewing your wireshark packet log, your VOIP media engine app appears to be transmitting its RTP packets two times (2x) for each 20Ms audio frame. Why is this?
I hope this information helps.
Randal
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speedvoip
Vetran
Joined: August 07 2008
Location: Canada
Posts: 156
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| Posted: April 22 2009 at 10:16am | IP Logged
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Hi Randal:
Thanks for your valuable info for issue troubleshooting. As what u said, I am very wondering why singleline phone app transmits RTP two times. Also maybe jitter is caused by real network congestion.
George
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support
Administrator
Joined: January 26 2005
Location: United States
Posts: 1666
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| Posted: April 22 2009 at 1:45pm | IP Logged
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Hi George,
Did enabling jitter compensation for the specified time duration help clean up the received call audio? It should have helped.
We have software updates that are scheduled for this spring/summer that will improve call audio due to packet loss, receive jitter and out of sequence RTP packet processing. I don’t have a hard date for these releases but as we get more improvements and new functionality implemented, we will release as soon as possible.
It might be a good idea to test one of these upcoming releases against your network. Your wireshark packet log will help us from the standpoint of having a real world test case to verify against the new software.
It is so busy here……
Randal
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speedvoip
Vetran
Joined: August 07 2008
Location: Canada
Posts: 156
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| Posted: May 05 2009 at 11:08am | IP Logged
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Hi Randal:
Very long not to come here. Just back from business trip. Applying for jitter compensation indeed do make a little improvement on callee audio quality.
Nice to get you plan to take software update on audio quality. Hope to join your beta program for testing at our real network world. Give wireshark packet capture log for feedback to u.
When update is available, ple hit me.
Thanks,
George
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support
Administrator
Joined: January 26 2005
Location: United States
Posts: 1666
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| Posted: May 05 2009 at 11:42am | IP Logged
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Ni hao George,
Yes, improving the jitter compensation and the packet loss concealment on the RTP receivers is something we will definitely get done. The planned changes will definitely help us all and most likely greatly improve your receive audio characteristics. We will use your RTP packet loss and timing info from your Wireshark log as a test case.
Let’s plan on working together again when the software update is available. I look forward to it.
I will post again when we have stable code.
Randal
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LanScape Support Forum -> LanScape VOIP Media Engine™ - Technical Support
Topic: Jittered Voice Quality
