[PLUG] VoIP on DSL?

[Kyle Hayes]

On Sunday 11 April 2004 11:56, Aaron Ten Clay wrote:
> Jeme is correct. VoIP connections generally consume around 55kbps
> after compression. (Both up and down, of course) So with an ADSL
> connection of, say, 1500/256, you could easily have 4 VoIP "lines".
>
> As to latency: most VoIP providers offer routers supporting QoS,
> preconfigured to pre-empt VoIP traffic, ensuring the lowest
> possible latency.

Uhm, this certainly doesn't dovetail with the experiences I had 
working for a VoIP firm for four years.

Different codecs have different bandwidth requirements.  G.723.1 is 
one of the lowest with around 7-8kbps.  G.729 is better quality but a 
little higher bandwidth.  About 12kpbs IIRC.  This is just for the 
codec audio stream, not other overhead!

If you sample voice at 8khz with 8-bit samples, you get standard G.711 
(which is no compression at all) and that is 64kbps.

There are a couple of more open codecs such as Speex and GSM.  Those 
are fairly good.  G.729 licensing is ugly.  $1 per port in single 
configurations and $10/port in multiple port configurations.  Yuck.  
G.723.1 is pretty bad too.

On top of the codec bandwidth, you've got packet overhead.  Generally 
you want to put two or three packets of audio into each UDP packet to 
reduce that overhead.  Too much audio data per RTP/UDP packet and 
you've got bad latency again and you'll need to bump up the jitter 
buffer.  Then you're in latency land again.  Over 200ms and the human 
ear can easily detect the latency.  Under that, and generally you 
don't notice it.   Note that using a microphone/speak ensemble 
through a sound card is a guaranteed way to introduce a _lot_ of 
latency.

I've personally run two simultaneous bidirectional calls through a 
28.8 modem without problems other than the horrible latency from the 
modem.  That was with G.723.1, 60ms frames, one frame per RTP packet.

If you are using G.711 and not doing any compression, then you'll be 
able to get one call per ISDN channel equivalent (i.e. 64kbps).  
Since there are free codecs, why would you want to run uncompressed 
audio streams?

So, let's assume that GSM plus packet overhead is about 20kpbs.  That 
leaves you with about a dozen simultaneous calls through a 256kbps 
link (you've got to count the lowest bandwidth side only).

Best,
Kyle