Improving VoIP

jivan

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A friend has a call shop and comes to me for advice. Fool.

How can he improve the quality of the calls? He uses VoipHit which is the best for the price, an unshaped SAIX line through WA and Linksys PAP2s.

He's been through every VoIP provider there is so he's happy with that. But what can he do. Maybe the QoS on the router would help so any advice there would be great.

Any advice would be most welcome.
 
How can he improve the quality of the calls?

Depends on where the calls are destined to?

an unshaped SAIX line

Is the line dedicated to VoIP only, what speed is it & how many concurrent calls are you trying to pump through it & what codec is being used?

Maybe the QoS on the router would help

QoS on ADSL is a waste of time, there are too many potential points of congestion that are beyond your control. Your only solution is overprovisioning bw relative to what you need.
 
Best quality will be using a local only account to a local VoIP provider on an ADSL dedicated for VoIP use only. Then the local VoIP provider will generally use proper dedicated international bandwidth for the international leg.

Count on 42.4kb/s ATM level bandwidth per simultaneous SIP/g.729 call over ADSL and try to stick within 75% of ADSL capacity.

Eg. on a 4mb/s ADSL, the upload is 512kb/s (sync / ATM speed). 75% of this is 384kb/s. Divide that by 42.4kb/s and you get 9 simultaneous calls.

NB. Do *NOT* use this link for DATA if you intend running anywhere near this volume of simultaneous calls.
 
Have you tried switching the WA unshaped Multi-Realm account between the Verizon and SAIX networks by appending .v or .s to the username to see whether or not there is any improvement in the call quality?
http://old.webafrica.co.za/support/multirealmunshapedfaq.html

You could also compare results using this voip tester to determine the jitter, packet loss and QoS. If the results are similar to those from a different unshaped ADSL connection then you may have to look at one of the more expensive solutions given above. You may need to run the tests several times to obtain representative averages.
http://www.voipreview.org/voipspeedtester.aspx
http://www.testyourvoip.com

Results for my 384Kbps line using shaped SAIX bandwidth from Telkom ...

Speed test statistics
---------------------
Download speed: 296912 bps
Upload speed: 107480 bps
Quality of service: 69 %
Download test type: socket
Upload test type: socket
Maximum download pause: 1100 ms
Average download pause: 170 ms
Minimum round trip time to server: 337 ms
Average round trip time to server: 342 ms

VoIP test statistics
--------------------
Jitter: you --> server: 7.4 ms
Jitter: server --> you: off
Packet loss: you --> server: 0.0 %
Packet loss: server --> you: off
Packet discards: 0.0 %
Packets out of order: 0.0 %
Number of supported VoIP lines: 1
Estimated MOS score: 3.9
 
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Thanks for the great replies.
Would a Linksys router be the same as it's also Cisco?
Most calls are too African countries, mainly DRC, Ethiopia, Congo, Pakistan....
It's on the G729 codec. I thought moving to 723 may improve it but after seeing gmza's great calculation I guess that this may not be a good idea.
The shop has 12 just expanded to 12 lines but this looks like it could spell trouble. Is 512kb the maximum upload speed telkom offers?
I'm going to run some tests now. I'll take a GB of semi-shaped fibre to see how it fairs.
 
Would a Linksys router be the same as it's also Cisco?
Cisco would be better, but the Linksys can probably do the job.

Most calls are too African countries, mainly DRC, Ethiopia, Congo, Pakistan....
Try finding a provider that has better direct interconnectivity to Africa/S.Asia

It's on the G729 codec. I thought moving to 723 may improve it but after seeing gmza's great calculation I guess that this may not be a good idea.
No harm in trying G723, it will save you some bw and reduce the probability of packet delay/loss.

The shop has 12 just expanded to 12 lines but this looks like it could spell trouble. Is 512kb the maximum upload speed telkom offers?
Well the real max upload throughput on a (error free) 4Mbps is about 440Kbps due to ATM cell overhead.

The G.729 codec will use a max of ~32Kbps per call, so 12 x 32 = 384Kbps throughput required. This however leaves little margin to compensate for any fowarding inefficiencies, so the chances of 12 concurrent calls being consistantly clear is highly unlikely.

Running G.723 will decrease your bw requirement to 12 x 24Kbps = 288Kbps, which should result in more consistant quality as there is a lower probability of encountering jitter & packet loss.
 
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Thanks for the comprehensive reply Roman4604. The ATM Cell Overhead was literally over-my-head. I'll have to read more (is there a course I can take or a book I can read).
I'll tell him to give a go at G723. Do you know of any providers that have direct links with African countries. The problem is that there is a market price that call shop clients will pay. Some will travel 30 min in a taxi to save 10c per minute. Even if the quality of the call is better.
 
The problem with G.723.1 is that terminating partners in Africa generally don't support it, so you're going to have transcoding happening at the carrier and quality degrades hugely with transcoding. You may find that where the route into Africa is a TDM interconnect (clear G.711 A-law channel) it works brilliantly, but for other countries where the route transcodes to G.729, it works terribly.

Thankfully, G.729 is almost universally supported natively, and a G.729 to/from G.711 A-law transcode will achieve a good MoS.

In reality, how often are all 12 phones going to be in used on live calls at the same time? Keep in mind that often people are busy dialling, ringing, getting engaged tones, hanging up, holding up a line without actually generating call traffic. This is particularly so in a call shop where people are in and out, paying, etc. It is quite likely that you may only peak at 9 or so live active calls at once.

Also, good luck getting the full 512kb/s to/from an international VoIP provider. I see there have been recommendations about bandwidth providers and, yes, Verizon / SAIX unshaped will give better results, but it is still extremely optimistic expecting such a consistent high-bandwidth throughput to/from an international provider.

If you are peaking with 12 calls (e.g. If you have 16+ phones), then you really shouldn't be stuffing around with this cheap nasty solution, but rather setting up a Trixbox/Asterisk PBX with IAX2 trunking to a local ITSP. This will let you easily carry 30+ (even up to 50+) simultaneous G.729 calls.

The problem is going to be finding an obliging local ITSP who can give you the rates you want and the IAX2 termination. You need to calculate the cost of multiple ADSL lines, unshaped vs local internet bandwidth, etc, vs the per-minute call costs. It may be cheaper paying higher rates, or it may be cheaper getting more expensive bandwidth and using two ADSLs...
 
@jivan: The current Voiphit rates to the destinations that you mentioned appear to be only marginally better than those being offered by Switch Telecom (gmza's company) so I would personally give them a try particularly if they are prepared to offer you some advice and support and hopefully improved call quality. You might also be able to arrange for additional services such as Voip-In numbers, voice mail, fax services, conference calls etc. The Betamax companies usually offer a no-frills service without support and payment and billing can often be problematic.

Voiphit rates (landline mobile)
DRC (R2.09 R1.86) Ethiopia (R2.05 R2.22) Congo (R1.09 R1.08) Pakistan (R0.72 R0.74)

Switch Telecom rates (landline mobile)
DRC (R3.06 R2.17) Ethiopia (R2.21 R2.07) Congo (R1.08 R1.08) Pakistan (R0.77 R0.77)
 
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No need for multiple lines or any fancy equipment when using ViBE.

http://voip.rheid.com/vibe/vibe.html

They have a simple NetGear router that can handle 30 concurrent calls with no quality issues. The ViBE software is what makes it work. It is a new form of QoS for real-time data and in its current incarnation is specifically designed for SIP traffic.

Have a look. We use one of their SIP trunks with ViBE and it works perfectly. We peak at around 17 calls (at times during the day) without any complaints or issues. All this over a 512kbps ADSL line with a IS capped account.

Without ViBE, we normally got to 4 or 5 calls and then the support phones started ringing with complaints for users. Since we moved over, no more problems.
 
Thanks for the info. I've contacted the makers voip-x.co.za for more information.

Sadly rheid's calling rates are extremely high, for example the DRC is twice as expense as my present provider, so how could I implement the ViBE system using my present setup to improve the voice quality.
 
Unfortunately, you are looking for the highest call quality at the lowest rate. That in itself is going to be difficult. In most cases, you will need to settle for one or the other.

From experience, the extremely cheap carriers are not too worried about your call quality, just high volumes. Most of these really cheap carriers are not going to use ViBE, since it adds to their costs. Good luck finding what you need. We settled on Rheid because of quality, not price. They are expensinve for certain destinations, but for local landline and mobile they are not bad at all, considering their high quality.
 
Well - we do have a solution.........

Hi,

As I have suggested before, I do believe that our solution gives a good balance between quality and price - we are generally on par or cheaper than voip-hit with the added advantage of local servers which gives a better latency and uses local bandwidth which is a lot cheaper. It also does away with the need to purchase through credit cards if buying from abroad.

We provide voip systems and routes to businesses currently - at higher rates simply because for that market Quality is the determining factor. However the cheaper rates we supply will still give a very acceptable quality. We will open up our servers to this shortly - If anybody would like to know more, send an email to [email protected] or pm me.
 
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Hi Jihan,

For one of the best local interconnects for National & International Termination.

Try:
http://www.amatoletelecoms.co.za/
[email protected]

They are a public licensed operator and offer very good QoS for VoIP interconnect, SIP or H.323.

For the VoIP ATA, I would recommend the AudioCodes MP-202. It is a 1, 2, 4 port FXS with optional FXO for PSTN fallback. With 2-port enet switch, with emb linux with router, firewall, VPN, VLAN, etc... and all QoS settings.
Try:
www.redlinx.co.za for stock.
 
3 factor
1 voip phone ,ata quality
2 network quality
3 land line quality
 
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