[gilkatho01]

I've been finding Skype incredibly useful for reducing my phone costs as I can call all of my required countries for 2.7c per min. Once I have my notebook connected to the wireless network I'm using Skype to make my calls, even when I'm in the office.

When in the office I use Skype even for local calls as quite often the call only lasts a couple of minutes and might cost 5c instead of our telcos 25c min.

I did get caught out recently though.

I called a supplier's call centre on their 13xxxx number using Skype and later discovered that Skpe charge 32 cents per minute for 13 numbers. I was onhold for most of it and the one call ended up costing ~$3 for the call which via our landline would have been 25c.

In summary, when travelling and at home I'll be Skyping as much as I can.

Wayne

[simongr]

I have tried skype. Audio wise it worked well between two skype users (i.e. using it like msn) but using it to call has been terrible - I am not sure if its a hardware issue - but the audio quality is almost unlistenable

[gilkatho01]

Occasionally I've had a bad call but in the main the call quality has been excellent. They are up to software version 2.5 so if you haven't tried for a while, give them another go.

I'm liking the "portable" functionality of taking my calls and phone account with me as I travel around.

[swissbignose]

Actually, they're up to version 3.0.

I use skype a bit for internal work calls and to call out, and have only experienced a problem where my bandwidth was constrained.

[NM]

Originally Posted by simongr

I have tried skype. Audio wise it worked well between two skype users (i.e. using it like msn) but using it to call has been terrible - I am not sure if its a hardware issue - but the audio quality is almost unlistenable

Skype claim to operate with a bandwidth between peers of between 3 and 16 kbps. Knowing that the audio stream uses UDP and RTP, and assuming a few features of their proprietary CODEC (such as 20ms samples and 3 samples per packet), the IP/UDP/RTP overhead is 40 bytes per packet sent every 60ms, or 5.3Kbps.

It is well known that latency has an affect on the subjective measure of call quality. The ITU has defined in G.114 that one-way latency for a voice call should not exceed 150ms. So a packet rate any less than 15/second will result in exceeding this recommendation in cases where the network latency is anything more than about 30ms, especially when receive jitter buffers are considered.

So this tells me the Skype bandwidth claim at the low end of their range is unachievable. And at the upper end of their range, that allows around 10kbps for the audio CODEC content.

The most common standard CODEC for low bandwidth VoIP calls is G.729 which has a network bandwidth requirement of 29.4Kbps including IP/UDP/RTP header overhead, or 8-16Kbps raw CODEC bandwidth.

So Skype is getting more than 40% better compression than G.729 CODEC, which I suggest is a direct result of the voice quality you have experienced compared with standard VoIP implementations. Skype determines the available bandwidth between the peers and configures its CODEC accordingly. So if you have limited bandwidth between the peers then expect poor audio quality.

I have serious reservations about the peer-to-peer operation fo Skype, especially when connected to the Internet without a NAT gateway. The Skype network relies on leveraging the resources of the distributed community of users for its directory services. I am unwilling to allow my computer and Internet bandwidth to be used for that purpose. However, I do recognise that it can provide cheap call for people in some situations.

I use VoIP extensively when travelling as I can make calls from Australia and the USA from anywhere I have sufficient Internet bandwidth. But it involves a significant amount of corporate infrastructure to do that and certainly more than 3-16Kbps of bandwidth to maintain a call (more like 100Kbps by the tiime the IPSec tunnel overhead is added).

[Febs]

Skype is my best friend when travelling. I use a Bluetooth headset and can walk around my hotel room while speaking to girlfriend/friends/family in Australia. I've never had a problem with any hotel Internet connection - either speed, latency, or firewall-related.

It might be a bit more expensive than some other VoIP solutions, but it just works!

My only problem with it is the hassle I have expensing my Skype credit when I get back. Our expense department sees the word "credit" and associate it with pre-paid mobile phone credit, which they don't reimburse. Always gets through in the end though.

Cheers,
- Febs.

[adrian]

I use it quite a lot when I am overseas. I find it very good to call landlines, but not so great to call mobiles. I can live with that, for the cost. My current mobile has wifi, so I don't need to call thru my computer anymore - it is just like making a mobile call, but without the cost. Very handy....

[mainly tailfirst]

Originally Posted by NM

Skype claim to operate with a bandwidth between peers of between 3 and 16 kbps. Knowing that the audio stream uses UDP and RTP, and assuming a few features of their proprietary CODEC (such as 20ms samples and 3 samples per packet), the IP/UDP/RTP overhead is 40 bytes per packet sent every 60ms, or 5.3Kbps.

IIRC, Skype quotes BW usage at 3-16 kBytes/sec, not kbps. <checks web>
Yep, http://www.skype.com/help/faq/technical.html confirms this.

I've never instrumented the stream (though I may be motivated to do so now that I intend to use it more), but from experience I get excellent Skype-to-Skype audio quality at between 5-6kB/sec. I find 3-3.5kB/sec yields acceptable quality even for trans-pac calls into a landline.

I don't know what Skype uses for POTS termination - I suspect G.711 or G.729 - so even the lower bound of 3kB/sec should be OK.
<snip>

So this tells me the Skype bandwidth claim at the low end of their range is unachievable. And at the upper end of their range, that allows around 10kbps for the audio CODEC content.
The most common standard CODEC for low bandwidth VoIP calls is G.729 which has a network bandwidth requirement of 29.4Kbps including IP/UDP/RTP header overhead, or 8-16Kbps raw CODEC bandwidth.

If we assume an IP packet size of 160B ('typical' number I remember from somewhere), 40B header o/head and packet every 30mSec, we get approx 5.2kB/sec - which dovetails nicely with my very rough measuring (add another 14B for Ethernet header per packet). So having an audio stream at ~3.8kB/sec (30kb/sec) explains alot why the quality is so good.
As you state above, to get the quality that they do with an 8th of that bandwdth would be extraordinary - obviously they don't.
I don't think Skype's audio codec is that brilliant, certainly I would expect that Motorola and Nokia et al. have better ones in their phones.
<snip>

I have serious reservations about the peer-to-peer operation fo Skype, especially when connected to the Internet without a NAT gateway. The Skype network relies on leveraging the resources of the distributed community of users for its directory services. I am unwilling to allow my computer and Internet bandwidth to be used for that purpose. However, I do recognise that it can provide cheap call for people in some situations.

To me, this is the real magic of Skype. I give up some bandwidth (usually an ammortized cost if I'm on a decent monthly plan), minor hit to processing (free) and in turn I get excellent sound quality calling most people for nix. Fair enough swap - almost a 17th century commons in action.

Aside: I never thought I'd have such a good tech thread on AFF. But there you go

mt

[NM]

Originally Posted by mainly tailfirst

IIRC, Skype quotes BW usage at 3-16 kBytes/sec, not kbps. <checks web>
Yep, Skype Technical Frequently Asked Questions confirms this.

I've never instrumented the stream (though I may be motivated to do so now that I intend to use it more), but from experience I get excellent Skype-to-Skype audio quality at between 5-6kB/sec. I find 3-3.5kB/sec yields acceptable quality even for trans-pac calls into a landline.

Ahh, yes. That makes more sense. Making it 24-120kbps.

But with VoIP, the most common cause of quality issues are dropped packets and jitter. Unfortunately there is no way to control those things on a public network like the Internet.

Another major factor for any softphone can be the audio device being used. Using built in microphone/speakers or even just adding a microphone and headphones to a computer generally results in poor audio quality. Using a USB audio device with its own DPS generally results in better quality. I use a Nortel USB DPS headset and get very good audio quality with various soft phones that I use, yet using a microphone and headphones with the same computer and software results in very much sub-standard quality.

Originally Posted by mainly tailfirst

I don't know what Skype uses for POTS termination - I suspect G.711 or G.729 - so even the lower bound of 3kB/sec should be OK.

I believe it is neither. Their direct POTS connection in most cases will be T1/E1 (ISDN in most cases) which use a 64kbps PCM which is similar to G.711. The connection from the Skype client to the POTS gateway uses the Skype proprietary CODEC which is neither G.711 or G.729 (not that it really matters, so long as it works).

Originally Posted by mainly tailfirst

If we assume an IP packet size of 160B ('typical' number I remember from somewhere), 40B header o/head and packet every 30mSec, we get approx 5.2kB/sec - which dovetails nicely with my very rough measuring (add another 14B for Ethernet header per packet).

G.711 CODEC results in 160 bytes of data per packet. Sample rate is 8000 samples/second using logarithmic PCM. The CODEC sample interval is 10ms resulting in 80 bytes per sample interval, and two samples are carried in each voice payload resulting in 160 bytes every 20ms.

Add to the 20 bytes for IP, 8 bytes for UDP and 12 bytes for RTP and we have 40 bytes of packet overhead or 200 bytes/packet at a rate of 50 packets/second. This results in 80Kbps requirement plus the layer-2 MAC requirements (ethernet, PPP, FR etc). On ethernet it grows to 87.2Kbps and on FR it would be 82.8Kbps.

G.729 is a little different, producing 10Bytes or data every 10ms sample and again sent as 2 samples every 20ms resulting in 20 bytes of data with the same 40 bytes of IP/UDP/RTP overhead. These 60 byte packets are sent at a rate of 50 packets/second resulting in a requirement of 24Kbps plus layer-2 overhead. On ethernet it grows to 31.2Kbps and on FR it would be 26.8Kbps.

Originally Posted by mainly tailfirst

So having an audio stream at ~3.8kB/sec (30kb/sec) explains alot why the quality is so good.
As you state above, to get the quality that they do with an 8th of that bandwdth would be extraordinary - obviously they don't.
I don't think Skype's audio codec is that brilliant, certainly I would expect that Motorola and Nokia et al. have better ones in their phones.

You have to wonder why they didn't just use G.729. I guess by keeping it proprietary they have to maintain control of the product/service??

Originally Posted by mainly tailfirst

To me, this is the real magic of Skype. I give up some bandwidth (usually an ammortized cost if I'm on a decent monthly plan), minor hit to processing (free) and in turn I get excellent sound quality calling most people for nix. Fair enough swap - almost a 17th century commons in action.

Providing you have a good ISP connection and don't suffer congestion, lost packets or jitter, and you use a decent DSP audio device on the PC, quality can be good. I still worry about using any Peer-2-Peer application on my PC. But I probably make 15+ hours of calls a week using various VoIP services, all via my home ADSL Internet service. But mine have the added overhead of running through an IPSec tunnel as well, adding a further 52 bytes/packet overhead so my G.711 connections use closer to 120Kbps and G.729 is more like 70Kbps

Originally Posted by mainly tailfirst

Aside: I never thought I'd have such a good tech thread on AFF. But there you go

mt

Its amazing what snippets of trivia can be find in a community like this .

[serfty]

What they said!