Can Voip replace your Telkom line??

Daveogg

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Their have been a few threads asking if voip can replace your normal telkom line this is my 2c on the question (sorry its a bit long).

As a background let me say i am a bit of a voip junkie. It’s a technology that I use daily to keep in touch with some very special friends in New Zealand.

If we look at telkoms recent pricing adjustments it is clear the effect voip competition is having on bringing down international call prices. 3 yrs ago a telkom call to NZ cost close to R4.00 per minute, now R1,50 . We know our favorite telecom company does not reduce prices because it wants too, only because it has too!!

So can Voip be as effective for national calls as it has been for international?? Can one effectively ditch your telkom line for a voip service??

Unfortunately at the moment I don’t think so. Let me tell you why.

I recently acquired a linksys spa 3000 analogue telephone adapter(ATA). For those who don’t know this piece of hardware, it connects to a broadband router, a telephone and telkom phone line. It allows you to place Voip calls (sip based) on your broadband connection to any voip service you have subscribed too. It also allows you to place calls via the telkom line. You can set up “dial strings”, effectively dialing rules, so the ATA knows how to deal with each call.

Although as I said I use voip daily, I am no expert on how it actually works. I have done my share of googleing and this is what I understand.

When I dial a New Zealand number, my ATA knows that the call is international(because it is 12 digits), it then passes the number to my international voip provider (currently voipdiscount.com). Voipdiscount has set up, or has access to a POTS (plain old telephone service) gateway for each country you may want to call. The ATA and the gateway, under the direction of the voip provider’s sip server, connect too each other and my call is connected.

Simply drawn the connection is:




Telephone ----ATA-----------------------------------------Gateway-------------------NZ Phone
......................................./........................ \
....................................../........................... \
........................International bandwidth........Connection controlled by
...................................(SAT 3)...........................sip server


So far so good, but what about the quality of the call. For me the test of the quality of the call is if the person I am calling notices anything different about the call, then it’s not good enough.

Call quality is a combination of “clarity” and “latency”.
Clarity is determined by the codec used and whether any voice packets are lost. Even low bandwidth codecs can achieve quite acceptable levels of clarity eg the Gsm codec used by cell phones only uses 13kbps * 2.

Latency is the Voip killer in South Africa. Remember when making a voip to pots call the traffic is routed through the gateway. From what I can establish these gateways are usually situated in Europe or the USA. This means to call a South African landline your voice packets must travel from your ISP (saix) to the gateway and then return back to South Africa before the person you are calling can hear what you said. His reply must do the reverse.
Ideally the latency between you talking and your speech being heard should be less than 250ms (effective latency). This is never going to be achieved making a “local” call with a gateway situated in Europe or the USA, where simple ping times are 400ms or more.

When Mweb announced that they had a voip service I hoped this meant they would set up a local gateway, so that local calls would remain “local” and the latency issues would be resolved.
I somewhat reluctantly signed up – as the cost saving hardly seems worth it – and have been testing for the last few days. Initial results are not that promising, there is still noticeable latency.

I have devised a way to determine what the effective latency is. Essentially what I do is make a call from my Voip line to my home phone and record both. I make a sharp noise (actually I just hit a bic pen against my keyboard), the resulting noise is picked up as it is made and then again when it is heard in the telephones ear piece. I then open that wav file in a sound editing application and measure the time between the sound impulses, which then gives me the effective latency.

Take a look here for the results. http://img222.imageshack.us/img222/3590/new10uf.jpg

Voipdiscount has an effective latency of 780ms and 950ms using a unshaped / shaped account respectively. Despite decent “clarity” of speech this is clearly not good enough to replace your land line despite call charges of only R0.08 / minute.

Mweb voip has an effective latency of 620ms. This was on both a shaped and unshaped saix account. This confuses me (believe me it does not take much to do that) if the gateway is local I would expect lower latency, on the other hand if international I would expect a difference of about 100ms between shaped and unshaped accounts.

I have subsequently tested skypeout to South Africa, which returned a latency of 750ms on an unshaped account – so similar to Voipdiscount.

So in conclusion presently I have a “voip” phone connected to my ATA. I use it for all international calls – which thanks to a weird business model by voipdiscount are all free.I also use it for friends and family who will tolerate my attempts at giving telkom the finger and will put up with some latency, but if I have an important call I use Telkom.
 
So bottom line: use VOIP to make international calls and not local calls
 
Daveogg said:
...
Latency is the Voip killer in South Africa. Remember when making a voip to pots call the traffic is routed through the gateway. From what I can establish these gateways are usually situated in Europe or the USA. This means to call a South African landline your voice packets must travel from your ISP (saix) to the gateway and then return back to South Africa before the person you are calling can hear what you said. His reply must do the reverse.
Ideally the latency between you talking and your speech being heard should be less than 250ms (effective latency). This is never going to be achieved making a “local” call with a gateway situated in Europe or the USA, where simple ping times are 400ms or more.

When Mweb announced that they had a voip service I hoped this meant they would set up a local gateway, so that local calls would remain “local” and the latency issues would be resolved.
.....

The quality of VoIP is nothing to do with the location of your service provider's gateway/proxy server/register server.

FYI,
http://en.wikipedia.org/wiki/Session_Initiation_Protocol

SIP is a P2P protocol (same as Skype, but proprietary), the actual voice or video content is carried by Real-time Transport Protocol (or RTP) between two ends which involved in communication. the proxy server (register server for Skype) offered by your service provider only helps you to find the one you want to call.

for PC-PC, easy to understand, the voice quality depends on the route between you and your callee (if you know you callee's IP, you can call her/him directly without any help).

for PC-phone, there will be a server (breakout server - route your call to tranditional phone system) in between, so the quality of voice will depend on the link between you and breakout server and the server's load/performance.

any VoIP provider, as long as you want to route the call to tranditional phone system, will deal with local phone company like Telkom here.

now, it's not difficult to understand why the VoIP quality to SA is not good and the rate is high (similarily, VoIP to cell is much higher than landline).

fortunately, there are something happening recently. the landline rate to SA has been dropped rapidly by some service provider, eg. VoipBuster 1 eurocent per minute, the quality seems improved
 
Hi renm.

I think we are saying the same thing - what you call the "breakout server" I call the gateway server. If you look at my diagram above you will see that my ATA makes a client/server connection to the gateway under the control of the SIP server.

As you rightly state the quality of the call is dependant on your connection to that server, I believe the quality of local voip calls will only become acceptable once that server is situated locally - thereby not requireing your voice packets to traverse SA-International links.

Where i dont agree with you is that Voip provider is dealing with Telkom(I guess Telkom refuses to deal with them). In call termination services you get white routes(legal routes basically dealing with the Telco companies) and Grey/Black routes where at least one end of the route is an ilegal or irregular connection into the telco network. http://www.answers.com/topic/white-route

I plugged my Voipdiscount log in details into a softphone and made a SA local call. Using netmonitor the softphone make a connection to 194.221.62.171 a reverse lookup shows this is the netherlands server of "TVI Connect".
http://www.tviconnect.com/

This is a bulk call termination service. I doubt they would then make a circuit switched call to my local number, but probably feed my call into another voip system which has a grey termination somewhere in SA.
 
Daveogg said:
Where i dont agree with you is that Voip provider is dealing with Telkom(I guess Telkom refuses to deal with them).

VSPs will (have to) deal with Telkom directly or indirectly. Telkom has got it's own VoIP service, this service can resell to small VSPs.

According to your guess, if those white/grey/black terminate servers are sitting outside SA, who is going to pay Server-TelkomSA long distance costs?

Daveogg said:
I plugged my Voipdiscount log in details into a softphone and made a SA local call. Using netmonitor the softphone make a connection to 194.221.62.171 a reverse lookup shows this is the netherlands server of "TVI Connect".
http://www.tviconnect.com/
any SIP device will contact with it's register server periodically to keep alive/presence (except IP-IP connection), but that will not affect the voice quality much.
strictly this includes register/proxy/stun server.
Daveogg said:
This is a bulk call termination service. I doubt they would then make a circuit switched call to my local number, but probably feed my call into another voip system which has a grey termination somewhere in SA.

No matter how your call has been routed, at end it has to be any grey/white/black server locally, that can explain why sometimes the quality is very bad (route passed through too many hops and some servers probably are just home setup/low performance.
 
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renm said:
No matter how your call has been routed, at end it has to be any grey/white/black server locally, that can explain why sometimes the quality is very bad.


The question i was trying to answer is whether VoIP can replace your telkom line and at present due to quality issues(mainly latency) i dont think it can. The reason for this latency is precisely how the call is routed.

What we need is Icasa to come to the table and force Telkom to interconnect at a resonable price to local voip providers.

It is interesting to note that Telkom’s unwillingness to interconnect with MWeb means that local calls from MWeb’s new VoBB service is more expensive than Telkom rates.

These interconnection problems mean that standard landline users will not be able to make calls to the MWeb 087 numbers. Interestingly enough it is possible to make calls to the 087 numbers from international destinations. MWeb however said that they are in ongoing negotiations with Telkom regarding this issue.
 
Reason for the high VoIP latency to SA landlines is cause the closest place you can interconnect with Telkom (indirectly) is in London, so the roundtrip is still up & down SAT3.

Interconnect to Telkom within SA is coming in the near future, however providers are going to find it time consuming to wade through Telkom's red tape.
 
Just to add another cents worth...

Latency and Jitter are the probably the factors that most directly effect the perceived voice quality.

in general, to be comparable to a landline call you'd expect single path latency of around 70ms, remember that a ping is a round trip time, so divide the ping by two to get a the single path latency.

The next problem is jitter, this is the time between packets arriving, the internet (being best effort) can deliver individual voice packets using different physical paths resulting in voice packets being delivered out of order, late or not at all. In order to reduce this as much as possible providers use something called a jitter buffer to reorder data packets and wait for late packets of data.

The larger the buffer, the better the quality, however this effectivley increases latency by however large the jitter buffer is.

a jitter buffer of 75 to 80ms is not uncommon, meaning total latency is somewhere around 150ms, above this it will become noticable and start sounding like an international call.

for reference a 64kb/s leased line from sandotn to cape town for instance has a latency of around 20 to 25ms adding a jitter buffer of 80ms gives 100ms path delay. it quickly becomes abvious how adding another 180ms, the average I see on internaional ADSL access can mess with the quality of the call.

The codec you use also affects latency and quality, some codec's like G723.1 and G729A offer bundled packets, where multiple voice packets are sent in a single UDP IP packet in order to reduce bandwidth utilization, this can save several kb in bandwidth however once again this increases delay and adds risk of losing significant portions of voice if the network drops your packet as you could drop 100ms or more of voice in one packet.

EDIT
As an example, 8K g729A/bundled (2 payloads per packets) has a packet rate/size of 50pps/21 bytes or about 21kb/s bandwidth whereas 8K g729A/bundled (5 payloads per packets) has a rate/size of 20pps/51 bytes or around 13.5kb/s a significant saving.

However if a udp packet is dropped on 2 payloads it translates to a 2ms loss in voice 1s/50 on 5 payloads 5ms of voice is lost 1s/20, so it becomes easy to see that if the network drops 20 packets due to congestion, on 2 payloads only 40ms is lost on 5, an entire second of voice is lost.

dunno if anyone finds this interesting, but parhaps someone finds it usefull one day.

D
 
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Thanks for the use full comments on latency
"Voipdiscount has an effective latency of 780ms and 950ms using a unshaped / shaped account respectively. "

Does any one have any other comment on UNSHAPED service? I see many posts in the forum on Telkom shaping. Will taking an unshaped service definately give better VoiP quality ?
Any comments

thanks Steve
 
i've spoken to guy who's part of a company that has developed a gateway system for telkom to use, its been ready to roll for around 6 months. Telkom is purposfully delaying its implementation because ICASA hasn't forced them to use it yet..
 
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