SAIX vs IS - Skype (and WoW) FEEDBACK

L

LethalChicken

Expert Member
Joined
Sep 14, 2004
Messages
1,340
Reaction score
3
Location
00-07-E9-13-97-54
Hi All,

Just thought I'd share some experiences with my skype experiments.

I got a test IS account yesterday and have tested Skype with both my IS and SAIX connections.

Both accounts are on standard shaped accounts with no proxies/VPNs

The tests were done at approximatelly these times:

Friday 16:00, Friday 23:00, Saturday 10:00, Saturday 15:00, Saturday 23:00

As you can see I didn't test during peak times in the week as yet, but my skype needs are after hours anyway.

The tests were to the echo test, to my family in pretoria (from JHB) and to a friend in london (he is South african so he understands the pain that is telkom).

SAIX:

Pro: Consistant Connection, no disconnects. Connects imidiatelly.
Con: Quality of call.

Me hear them: Ok, some words got lost or garbled. (score: 3.5/5)
Them Hear me: the calls @ 23:00 were semi decent, the other ones the people could not hear me. (score: 2/5)

IS:

Pro: Quality of call.
Con: Struggled to estabilish connection, had a total of 3 disconnections during test period.

Me hear them: Occasional loss of quality, but still can understand all (4/5)
Them hear me: Almost Perfect (4.5/5)

One thing I noticed that with IS you have to let the person finish talking before you do it's not full duplex, with saix it's more full-duplex :)

Conclusion:
Even though SAIX lets handles 2 people talking at the same time better than IS, and there are less disconnections, the IS quality is MUCH better and all the conversations were completely understandable.

As a side note, both connections were MUCH better with a VPN, but if like me you can't afford the VPN, IS is DEFINATELY the way to go.

As far as WoW goes, the results were much clearer to describe:
I play on the US Earthen Ring Server.

SAIX: Quicker responses, Very jumpy and big lag spikes. (lag meter shows 600-800ms, in IF 2000-3000ms)
IS: Very smooth no jumps, but a steady 1-1.5 second lag (lag meter shows 1000-1500ms, in if 2000-3000ms)

Conclusion here, neither one is good, but the packet loss on SAIX (jumping) makes the play difficult and unpredictable, on IS if you can get used to the 1sec lag and plan accordingly, you should be ok. ( I did not try raidning as yet)

Once again with a VPN, it would all be much better.

Well hope it helps, please post if you have any test experiences.
 
@blobnow : thanks for your feedback mate. I would however like to point something out to you for consistency purposes. Skype uses a node -> supernode approach to p2p. In laymens terms, the node daisy chains itself from a supernode, and the supernode acts as a "switch" for the calls. To become a "supernode", you need to have a legal IP address and the relevant ports open on your firewall. Now, why I mention this : If you are firewalling yourself, you will not be a supernode, so your connection will daisy chain itself. Given that most people properly firewall ADSL connections, its rather safe to assume that most ADSL users are not switching calls as supernodes. Now, dialup connections are a different story - alot of people still dont firewall them, so they have become natural candidates for supernodes. The implications of this : even though you are on ADSL - you could still be routing via a dialup connection.
My point? When you do tests like this, it is imperative that you make sure that on both networks, you are a supernode. You can do this by placing the machine you are doing the tests from on your DMZ, or (if you are brave) dont firewall the connection at all. Alternatively, you could use port forwarding on your router (with some translation) to make sure that your machine is seen as a legal IP with the relevant ports open.
As a supernode you stand a MUCH better chance of getting a good Skype connection. I have both SAIX and IS accounts, and the quality is equally good all the time on both networks - but I have made sure that I am always a supernode.
A classic example : Vodacom 3g network. If you use the standard "internet" APN [private IP], voice quality is OK on skype. If you use "internetvpn" APN [public IP] with the relevant ports open on your local firewall, voice quality on skype is excellent.
I am of course assuming that you havent done this all already... If you have, ignore :D
Cheers ~
ScrnScrm
 
Just an idea but have you considered enabling Skype's "Display technical call info" in the advanced options? That way you'll get a better idea about packet loss etc.
 
I didn't know about the supernodes ... so I didn't take that into account, my main focus was 'normal usage' ... which is no sepcial setup, connect skype and talk.

I tried to base the tests on the 'out of the box' approach where no setup or specific configration would be needed.
Also wouldn't the fact that I was using the same setup on both networks balance the tests? I'm guessing from the description if I changed the setup I would get better connection, but if I'm comparing apples with apples here, it should be a valid test.


Also as a matter of interest, if I set myself up as a supernode, wouldn't that eat up a lot more of my bandwith ?
 
Last edited:
BlobNow said:
I didn't know about the supernodes ... so I didn't take that into account, my main focus was 'normal usage' ... which is no sepcial setup, connect skype and talk.

I tried to base the tests on the 'out of the box' approach where no setup or specific configration would be needed.

Also as a matter of interest, if I set myself up as a supernode, wouldn't that eat up a lot more of my bandwith ?
Click to expand...

Yes it would eat up more bandwidth because you will be routing calls for other users. Thats why I only enable the port forwarding when I need to make a call...
 
BlobNow said:
Also wouldn't the fact that I was using the same setup on both networks balance the tests? I'm guessing from the description if I changed the setup I would get better connection, but if I'm comparing apples with apples here, it should be a valid test.
Click to expand...

Not necessarily. There are other variables. IS have far fewer dialup users than Telkom, so naturally there will be fewer supernodes... Thats why, for consistency, I suggested you make yourself a supernode. That way you can rule this out of the equation and get true apple for apple comparisons...
 
I tried the show advanced settings, and I see that even though the roundtrip on SAIX is about half of IS, the packet loss on IS is about 0.2%-0.9% and on saix it's hovering between 12%-15%. And packet loss is much more of a criminal then latency
 
ScrnScrm said:
Not necessarily. There are other variables. IS have far fewer dialup users than Telkom, so naturally there will be fewer supernodes... Thats why, for consistency, I suggested you make yourself a supernode. That way you can rule this out of the equation and get true apple for apple comparisons...
Click to expand...

Fair enough, but I'll leave those tests to brave people with more bandwith then me ;)
 
hehehehehe. in most countries, people would be worried about security. in south africa, we are worried about the precious bandwidth wastage :D. hehehehehehehe. how fl@cked up is that :D
 
well I'm also a bit paranoid about security ... I have a nix box for the firewall and I have a firewall, a trojan scanner and an antivirus on each windows pc ;)

But I'm still so worried about international operators as well as telkom all destroying VOIP.

Internet prices going up, phone calls going up, petrol going up and in december, interest rate going up .... and my blood pressure follows ...
 
You must log in or register to reply here.
Back
Top
Sign up to the MyBroadband newsletter
X