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SKYPE

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SKYPE

I am having a bad time with Skype. My friends say it works fine. Is it because I am on Plusnet BBplus? Would things improve if I moved say to premier?

CS
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Re: SKYPE

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I am having a bad time with Skype. My friends say it works fine. Is it because I am on Plusnet BBplus? Would things improve if I moved say to premier?


I have heard from a number of people that Skype frequently has quality of service issues. This is probably due to a combination of things:

1. Skype uses the bandwidth of other random Skype users to proxy your call (probably without their knowledge) if your NAT prevents it making a direct peer to peer connection with the other party. This means you're then subject to the quality of those other users' connections.

2. I believe Skype will fall back to sending voice traffic over a TCP connection if it can't make a UDP connection. This is a Very Bad Thing, TCP is wholley unsuitable for realtime streaming applications and will likely cause serious quality of service issues due to both it's slow-start design and head-of-line blocking.

Of course, as with all closed protocols, by choosing to use Skype you are locked into a particular vendor (Skype themselves) and you cannot change if you are unhappy with the service. You should switch to using an industry standard, open protocol such as SIP instead (which is what practically everyone except Skype uses).

The other problem you may well have is that your connection may well be in use by other software - the upstream side of an ADSL can manage about 30KBps - that means it takes about 50ms to transmit a 1500 byte packet (which is the normal maximum packet size) so just two large packets being put ahead of your voice data will cause 100ms jitter which will almost certainly be noticable. If you want to do other stuff over the connection at the same time, the only solution is to set up a router to do prioritised queuing of the network traffic, and setting that up certainly requires some technical knowledge. (I route my network traffic through a Linux machine which I have configured to do priority queuing for me, with that running I can pretty much hit my connection as hard as I want and phone calls going over it will remain unaffected).
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SKYPE

That is a fair point.

However (I know full well the security level risks associated with skype due to Nat traversal etc) the call quality on Skype is superb as long as you are not downloading at 170k and uploading at 30k!

As for plustalk it is simply rubbish even if you are doing absolutely nothing! I regret signing up for it!
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SKYPE

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the call quality on Skype is superb as long as you are not downloading at 170k and uploading at 30k!


Having not used Skype myself I can't comment on the call quality - I can only say that I've heard quite a few people complaining.

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As for plustalk it is simply rubbish even if you are doing absolutely nothing! I regret signing up for it!


I've not had any problems with PlusTalk's quality since I fixed the QoS queuing on my router. It's possible that Skype has larger jitter buffers than PlusTalk (PlusTalk appears to have a 100ms jitter buffer). Larger jitter buffers will protect you more from drop-outs caused by other traffic on the connection but will increase the latency.

My suggestion is to set up QoS queuing if possible (I can post the config for my Linux router if that would be helpful), or failing that try reducing your MTU down to somewhere around 500 bytes.

The other possibility is that you might be using the GSM codec rather than ULAW, which will lead to the connection sounding more "muffled", much like a cellphone call. If that's the case, try switching to ULAW or ALAW.
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SKYPE

rivendell2

thanks for those suggestions.

However, the Merthyr exchange is a REALLY good exchange and i really am literally less than 1 mile away from it and it was a former switching exchange!!!

So, in the grand scheme of things it is a good exchange and i am close to it!

I disagree with what you are saying. Try skype and you will understand. I have used all the codecs suggested and seriously the call quality is diabolical. can download at 200k+ and upload at 30k+ so it is not line problems!

I realise that sip is enhanced security and necessary so that is founded over skype. however, the codec i should use is chargeabvle GSMx29 or something i think.

Anyway, ALL other codecs are crap trust me! Plus Net should introduce a good codec of their own!
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SKYPE

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However, the Merthyr exchange is a REALLY good exchange and i really am literally less than 1 mile away from it and it was a former switching exchange!!!


I'm not sure what your local exchange has to do with anything?

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I disagree with what you are saying. Try skype and you will understand.


Both my PlusTalk and SipGate accounts give me quality that is on-par with an ISDN (as do my VoipUser DDIs) when I'm using the ULAW or ALAW CODECs.
Skype is not an option since it is a propriatory protocol and therefore incompatable with my equipment.

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can download at 200k+ and upload at 30k+ so it is not line problems!


I'm sorry but the connection requirements of a voice protocol are completely different to the requirements of a download protocol.
When downloading you require a high throughput connection - latency and jitter don't matter at all and low levels of packet loss will have only a small effect.
When transmitting realtime voice traffic you need a low latency, low loss and low jitter connection, but throughput doesn't matter so long as it is above the codec's data rate (around 64Kbps for ULAW or 13Kbps for GSM). On the upstream side of your connection (which is relatively high latency) just 2 packets out of place can introduce jitter exceeding 100ms. This would not be noticable on a bulk data transfer, but 100ms jitter will easilly cause the peer's jitter buffer to underrun and be audible as a noticable break in the audio.

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however, the codec i should use is chargeabvle GSMx29 or something i think.

Anyway, ALL other codecs are crap trust me! Plus Net should introduce a good codec of their own!


I'm not sure I understand what you mean.

GSM is the CODEC used by cellular telephones in most of the world - it reduces bandwidth usage to around 13Kbps by linear predictive encoding and so there is a noticable loss of quality. However, the loss of quality is considered acceptable for low bandwidth applications such as cellular telephony.

ALAW, on the other hand, is the CODEC used throughout most of the European TDM based phone network (i.e. the digital public phone network). ULAW is the CODEC used by the American telephone network - they are both equivalent to eachother and provide an 8KHz voice channel at 64Kbps, which is considered good enough for telephony (certainly better than GSM),

There are a number of other CODECs in use on VoIP systems of varying qualities - for example, Speex provides a 16KHz voice channel and is therefore considerably better quality than the public telephone system. However, if you are calling someone who is on the PSTN there is little point in using a wideband CODEC since the higher quality will be lost at the gateway.