VOIP breaking up

VOIP setup and troubleshooting
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Dazzled
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Re: VOIP breaking up

Post by Dazzled » Wed Oct 26, 2011 10:46 am

Variance is a statistical measure - eg, http://www.experiment-resources.com/sta ... iance.html and can't be configured. The ATA buffers are set in the firmware and are severely limited by the small available RAM in these devices.

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Re: VOIP breaking up

Post by Chalin » Wed Oct 26, 2011 3:55 pm

abbo56 wrote:I forgot to ask:
For us less technically savvy computer users, is there a way to adjust the jitter configuration to experiment with solving this problem?
Thanks
Hi abbo56,

We cannot make any changes to the jitter configuration on the connection from our end, but depending on the VoIP device that you use; you may be able to enable jitter buffering on the device.

Regards,

Chalin

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Re: VOIP breaking up

Post by Dazzled » Wed Oct 26, 2011 7:29 pm

Re Chalin's post above, Linksys comes to mind for configurable hardware. Computers have RAM to burn and you can adjust the RTP jitter buffer with some of the softphones. Linphone (default is 60 ms) is particularly easy to change. So is Ekiga.

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Re: VOIP breaking up

Post by abbo56 » Tue Nov 08, 2011 9:30 am

Hi Chalin,
I operate a Dynalink RTA 1046VW modem.
I cannot find any function that relates to jitter buffering. The nearest control that may be in the area to which you refer is "Quality of Service" in the advanced tab of the modem login. There are two sub sections labelled :"Bridge QoS" and "IP QoS" Would either of these be related to your suggestion?
VOIP issues still persisting. Will get access to a replacement VOIP capable modem on 26Nov11 to see if my modem has any personal problems
Regards

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Re: VOIP breaking up

Post by Dazzled » Tue Nov 08, 2011 9:37 am

The Dynalink is like most hardware models, and doesn't have a configurable setting. It is a very reliable ATA.

The easiest way to experiment is with a suitable softphone. QoS is designed to give voice priority over other traffic on your local network.

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Re: VOIP breaking up

Post by netless » Sun Nov 27, 2011 1:36 pm

I have been having exactly the same problem with Exetel VoIP since early this year - prior to that it worked fine. I’m on an 8Mbps plan and during the day I get about 3 – 6 Mbps and generally VoIP works fine. During the evenings my speed regularly drops to sub 1 Mbps – often lower, perhaps down to 0.3 Mbps. This daily fluctuation in speed between day & evening is as regular as clockwork. During the evenings I am also experiencing severe VoIP problems – just as you describe – outgoing voice appears OK, however incoming voice is broken to the point it is impossible to understand.

After logging a fault log, going through the usual testing procedures to eliminate the possibility that the problem is at my end – Exetel finally started to take the issue up with their supplier. At first the problem (network contention) was going to be fixed in a month, then two .... the last responses I had from Exetel support were

20 August 2011-11-27
“Please be advised that our wholesale supplier investigated the reported issue and confirming that this is due to contention at the local exchange. They have included this DSLAM in to the Upgrade list however there is no ETR on this.”

22 August 2011-11-27
“Kindly be informed that as per our supplier's information, this upgrade has been scheduled to be completed by end of December 2011 or beginning of January 2012.
We appreciate if you could bare with us until our supplier upgrade their infrastructure. “

So while I live in hope (perhaps forlorn) that the network contention issue at my exchange may be fixed by January 2012 – I am very pessimistic about much being done to keep the current ADSL network running while the NBN rolls out.

I think the best option is for affected users to keep logging fault tickets on this issue & keep this forum discussion alive to keep the pressure on Exetel and their supplier to fix this unacceptable level of service provision.

In the meantime I continue to live with an Exetel VoIP home phone service that is unusable most evenings.
abbo56 wrote:I too am having issues with VOIP
It started back about July this year and has continued.
At times (particularly in the evening from 5pm EST onwards) any VOIP call made by me, or from others calling in from an outside PSTN or mobile to my VOIP number, has the issue where I cannot understand the other person because their signal breaks up to series of staccato bleeps and hollow sounding echos. They are able to hear me very clearly. This problem can occur at other time in the day but is particularly prevalent in the evening. There are times during a call when the connection works fine and then it will intermittently become unusable. People calling in to my PSTN does not generate the issue
I raised a fault log and spent several weeks in discussion trying to solve the problem. Different modems were used and a soft phone was tried. All had the same issue
Down load rates were recorded and there were significant fluctuations. Poor readings coincided with poor VOIP
The codec was adjusted but no change.
At the end of all this I received the following email from tech support:
"I have investigated the issue with our wholesale supplier and it appears the slow speed matter (which affects your VoIP service) is due to network related problems from their end.

We have already requested an ETA on the matter from our supplier. I will be in contact with you, once we get hold of more details in this regard from our wholesale supplier." and

"Please note that ETA provided by our wholesale supplier is June 2012.

We apologise for the inconvenience caused."

So for those of you who are having similar issues it would appear there is no easy fix.
Because June 2012 is so far away I did ask if changing from ADSL1 to ADSL2 would fix the problem. The answer was indeterminate because of a lot of other technical possibilities that could affect the result.
When I signed up with Exetel I chose a 1500 plan so the high download rates would allow good VOIP but it would appear I am paying for a service speed that cannot be delivered.
I am a very light internet user. I really only need higher speed for VOIP..
Does anyone have any other ideas on how to get around this VOIP issue. And do you know if changing ISP will fix the problem or is the problem at the wholesale supplier end going to be common across all VOIP/Internet suppliers?
Thank you

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Re: VOIP breaking up

Post by netless » Sun Nov 27, 2011 4:55 pm

This really is starting to look like a jitter problem - more than a network speed problem per se.

I have just run a network packet capture test on a VoIP session with the following results

Source IP Drop by Jitter Buffer Out of Sequence Wrong Time Stamp
192.168.1.3 (me) 0.0% 0.0% 0.0%
58.96.1.2 (Exetel) 48.4% 0.0% 20.0%

The jitter buffer setting for the simulation was 50ms

This appears to be very consistent with what we observe - good voice out - broken voice in.

Some feedback from Exetel's network engineers on this jitter issue and solutions for its resolution would be greatly appreciated.

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Re: VOIP breaking up

Post by Dazzled » Sun Nov 27, 2011 5:46 pm

I don't know what the network people can do with an exchange and backhaul issue, but it is easy with some softphones to adjust the jitter buffer. (eg, Linphone is available for most operating systems; the adjustment is at Go->Preferences->Network->Number of buffered voice milliseconds. 60 to 512 ms is available. Ekiga for Linux and Windows has a similar adjustment under the Codec menu.). An experiment with a larger setting might be worth trying.

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Re: VOIP breaking up

Post by netless » Sun Nov 27, 2011 11:51 pm

Thanks Dazzled for the quick reply - unfortunately my NB9W Netcomm ADSL Router - driving my conventional radio phone system - does not appear to have any Jitter adjustment facility.

Let me know if this is not the case - I have already enabled QoS for UDP packets - but this only affects outgoing traffic - which is not the problem in this situation anyway.

Also, I've tried modelling different jitter settings with the packet data I have downloaded - it looks as if I would need jitter buffering of 200 - 300 msec to have much effect.

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Re: VOIP breaking up

Post by Dazzled » Mon Nov 28, 2011 7:06 am

As I mentioned above, few hardware ATAs have this facility, Netcomm and Dynalink included. Exceptions include Linksys and those that have had open-source firmware installed.

The audio part of a VoIP call uses the RTP protocol on a direct connection between computers. The UDP parts of the traffic are the control signals between you and the Exetel server.

My Optus-equipped exchange doesn't have this problem so I can't experiment, but the softphones mentioned have a computer's worth of RAM to play with and can be set to 300 ms to see what happens with incoming traffic. The default audio packet on Netcomms is 20 ms worth, but this only affects sent packets.

Domestic modems run a Linux system in remarkably little RAM - for the Netcomm, telnet 192.168.1.1 and type the command cat /proc/meminfo. The output headings are explained at http://www.redhat.com/advice/tips/meminfo.html. The application voice runs VoIP in this modem, with buffer sizes fixed.

There is a diagram showing the connections for a VoIP call from phone A to phone B at http://www.packetizer.com/ipmc/sip/pape ... _sip_voip/ There's a lot of info at that site, eg sections linked from http://www.packetizer.com/ipmc/papers/u ... index.html

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Re: VOIP breaking up

Post by netless » Mon Nov 28, 2011 4:17 pm

Hi Dazzled

I can certainly telnet into my nb9w as you suggest - however the cat command & many others ls, pwd, cd, etc are not available - the /proc/meminfo command is not there either.

I log in [admin/admin] to some default directory with a bunch of commands mainly related to setting features on the nb92 eg voice & some networking unix commands.

However I can not find any other standard UNIX directories eg /bin or the location of any standard UNIX commands [if they exist in the implementation] such as pwd, cd, ls etc to explore the UNIX structure.

Any suggestions?

netless

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Re: VOIP breaking up

Post by Dazzled » Mon Nov 28, 2011 6:53 pm

Sorry about that netless - I have some Netcomm routers to play with and check but not an NB9W. These things are so compressed that the firmware uses full path names. cat is usually available. Try something like echo /bin/* to see what's there. There will be a Busybox shell in there as well - try echo ; exec /bin/sh to try to get the menu page. There is a discussion of the older NB5 online at http://www.beyondlogic.org/nb5/ but I don't know of a published NB9W investigation. The makers don't publish manuals so you may have to stab about with Linux commands until it comes good. For an example of memory on a combined modem/router/ATA, I've got a bridged Dynalink running here at the moment, so I disconnected it and had a peek - MemTotal: 14148 kB, so it's pretty tight.

The Netcomm prompt at login lists the apps that have been put on board to do the job. Each has an argument summary.

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Re: VOIP breaking up

Post by netless » Mon Nov 28, 2011 7:52 pm

Thanks Dazzled

Another quick question if I may.

I downloaded the current version of LinPhone - but I can see no place in preferences to adjust the jitter buffer - perhaps you have an older version.

Can you let me know what your version number is?

Thanks

Netless

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Re: VOIP breaking up

Post by Dazzled » Mon Nov 28, 2011 8:11 pm

I haven't updated mine - it's Linphone 2.0.1 for Linux. I notice that an update is overdue. Linphone for Linux, after running once, creates a resource file, .linphonerc, where the default setting is audio_jitt_comp=60 in the RTP section. This is as well as the GUI adjustment. Info is at http://www.linphone.org/eng/documentati ... phone.html. PCMA is the only useful codec.

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Re: VOIP breaking up

Post by netless » Mon Nov 28, 2011 11:06 pm

Hi Dazzled

I have now downloaded Ekiga - this looks like a simple softphone - and as you suggest the jitter is adjustable in the Audio Codec.

I'll give this a go and see if it helps.

Thanks for you help

Netless

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