Choppy Incoming Voice During Calls

[boriz]

I have set up my PAP2T with all the recommended settings (g729a codec preferred) and put in all the right values for dialing rules etc.

1 out of 5 times during a call, the voice quality will be very bad. It will be rapid cutting out for incoming sound. Outgoing sound is fine though.

I have Rogers 10mbit/1mbit internet and there's no bandwidth being used during calls. Most of the time calling the person back will fix the issue, but it seems like sometimes my call is routed improperly to a broken station. Does this have anything to do with my settings and is there anything I can do about it? It happens quite often.

Looking through the forum history hasn't helped and the closest thread to what I'm experienced was this:
http://forum.freephoneline.ca/viewtopic ... audio#p822

My calls only have choppy incoming sound however.
Thank you for reading, I am quite satisfied with this service so far, and it would be nice if this issue was nonexistent!

[FONGO_kris]

Try turning the Jitter Buffer Level down, to High, or Low (Usually starts by default on Very High). Also try switching OFF "Use Preferred codec only"

[leper1983]

I seem to be getting this issue also.. its slightly annoying i have to admit

[FONGO_steve]

I seem to be getting this issue also.. its slightly annoying i have to admit

I know this question comes up a lot, but I have to ask; have you done the necessary port forwarding in your router?
Also, what connection speed to you have? Are you downloading/gaming a lot? Lots of computers on the network sharing bandwidth?

This is a good time to highlight the fact that more expensive internet routers are always a great way to go for VOIP. I own and use the D-Link DGL-4300 and DGL-4500 routers on two different home networks, both with heavy VOIP, download and game traffic. I've never once had to access my router setup to port forward, adjust QoS or anything of the sorts. Any D-Link router with their QoS engine in it should virtually eliminate the port forward hassle that many cheaper routers have. I know that Buffalo also makes a router with comparable automatic QoS control. I'm not familiar with Linksys.. but they're sure to have something comparable. So for those who are serious about adding VOIP onto their network, it doesn't hurt to spend a little more on your router instead of buying the sub-$100 model just because it's cheaper! A lot of this port forward hassle that is traditionally associated with VOIP can be virtually eliminated in this league of router.

[leper1983]

Yea i had all the correct ports forwarded but it was the defaults also.. which i then changed just to see if i still have issues.. plus i found this page here http://forum.freephoneline.ca/viewtopic ... 98&start=0 that had a few settings different then what i was using.. so far so good tho

[boriz]

I'll try finding this jitter buffer level setting and turning it differently when I get back to Toronto.
Will also turn off the preferred codec setting as well, that might help. Thanks for the replies everyone.

I have a pretty good netgear wireless N router (I believe it's a WNDR3300) and doubt it's the source of the problem. Right now there's no money available to experiment with a different router but maybe in time I'll get something more advanced.

[boriz]

I started changing the values on the jitter level but it's not making much of a difference. Also, this problem is more common, one every 3 calls gets the choppy voice. I can get a sample clip if you'd like.

[volenin]

0) go to http://myvoipspeed.visualware.com/, make sure your VoIP ports are routed to your desktop (the PC you are loading the test from) and see the VoIP line 'rating'. This should basically eliminate non-FPL related issues. If you see your results are good (run the test at least 10 times, ideally during different times of the day), then
1) see the stats on the 'dropped packets' on your router admin console (on the Status or Info page - don't recall for sure). If you see significant # of dropped packets, then
2) play with jitter levels

Good luck!

[boriz]

I get about 4.5~4.7ms average jitter and 0% packet loss (at the montreal server).

Been having some problems with my router today for some reason (it's forwarding port 80 to the pap2t instead of my webserver) so I turned off DMZ and that seems to be helping. I'll report again with a voice sample next time this happens. Thanks for your help!

[boriz]

Hello everyone, it seems I have fixed the issue!

Two weeks ago I did some reading on "recommended pap2t changes" and it was said that changing the RTP Packet Size from 0.03 to 0.02 would result in more packets and better quality... so I went for it and now I haven't seen the ringing problem in 2 weeks! Mission accomplished, just wanted to share with everyone else.

If you're not sure where that config section is, go to the SIP tab and RTP parameters, it's the second box down. Good luck!

[volenin]

Yep... Or go through the config guides posted on the other board (ATA Configuration Settings or smth like that) - it has this setting there explicitly set to 0.020. You might actually want to go through the setup ref. guide anyways - some recommended settings there might improve the call quality / reliability....

[FONGO_kris]

Yes, in regards to the 0.020: If your payload values for each codec are off, you may experience bad or choppy (or NO) voice quality.