A Broadband and ADSL forum. BroadbanterBanter

Welcome to BroadbanterBanter.

You are currently viewing as a guest which gives you limited access to view most discussions and other FREE features. By joining our free community you will have access to post topics, communicate privately with other members (PM), respond to polls, upload your own photos and access many other special features. Registration is fast, simple and absolutely free so please, join our community today.

Go Back   Home » BroadbanterBanter forum » Newsgroup Discussions » uk.telecom.voip (UK VOIP)
Site Map Home Register Authors List Search Today's Posts Mark Forums Read Web Partners

uk.telecom.voip (UK VOIP) (uk.telecom.voip) Discussion of topics relevant to packet based voice technologies including Voice over IP (VoIP), Fax over IP (FoIP), Voice over Frame Relay (VoFR), Voice over Broadband (VoB) and Voice on the Net (VoN) as well as service providers, hardware and software for use with these technologies. Advertising is not allowed.

Who decides which codec to use?



 
 
Thread Tools Display Modes
  #1  
Old May 26th 07, 03:14 PM posted to uk.telecom.voip
John Miller
external usenet poster
 
Posts: 15
Default Who decides which codec to use?

Let's say person A is using SIP provider A on an ATA, and person B is using
SIP provider B on another ATA.

Person A calls person B (both using SIP, but with "real" phone numbers, so
not via ENUM). Which devices are deciding (comparing their codec priority
list) which codec to use?

- Provider A and provider B? If this is the case, can it be that the link
between provider B and the ATA of person B can use another codec? Also for
the link of person A's ATA to provider A?

- Provider A and person's B's ATA? Is this case provider B is
"transparant"?

- Person A's ATA and person B's ATA? In this case both provider's priority
list doesn't matter?

Thanks!


  #2  
Old May 26th 07, 06:30 PM posted to uk.telecom.voip
Philippe Deleye
external usenet poster
 
Posts: 76
Default Who decides which codec to use?


"John Miller" wrote in message
...
Let's say person A is using SIP provider A on an ATA, and person B is

using
SIP provider B on another ATA.

Person A calls person B (both using SIP, but with "real" phone numbers, so
not via ENUM). Which devices are deciding (comparing their codec priority
list) which codec to use?


There are 2 possibilities I believe:

a) Provider A and Provider B have interconnection agreements, and provider A
is able to convert the "real" phone number of person B into a SIP adress.
In this case there is a SIP conncetion between A and B: here I assume Person
A's ATA and person B's ATA will ngotiate the Codec

b) Provider A and Provider B do not "know" each other: Provider A will built
a PSTN connection to Provider B's network.
In this case, there is no link between the codes used.
The connection between person A and provider A will use wathever Codec is
supported by A, and the connection between person B and provider B will use
wathever Codec is supported by B.

At least this is my opinion ...


  #3  
Old May 27th 07, 12:03 PM posted to uk.telecom.voip
Jon Farmer
external usenet poster
 
Posts: 40
Default Who decides which codec to use?

John Miller wrote:
Let's say person A is using SIP provider A on an ATA, and person B is using
SIP provider B on another ATA.

Person A calls person B (both using SIP, but with "real" phone numbers, so
not via ENUM). Which devices are deciding (comparing their codec priority
list) which codec to use?

- Provider A and provider B? If this is the case, can it be that the link
between provider B and the ATA of person B can use another codec? Also for
the link of person A's ATA to provider A?

- Provider A and person's B's ATA? Is this case provider B is
"transparant"?

- Person A's ATA and person B's ATA? In this case both provider's priority
list doesn't matter?


There are a number of answers to this question depending on how provider
A and B have configured their network and how they interconnect.

As a general rule whatever you are directly conencting to i.e UA, B2BUA,
media relay will negoitate a codec that is common to both you and them.
It is possible that the interconnect, if used, will use a totally
different protocol and will be performing transcoding to enable this.

HTH
Regards

Jon
  #4  
Old May 27th 07, 12:03 PM posted to uk.telecom.voip
Jon Farmer
external usenet poster
 
Posts: 40
Default Who decides which codec to use?

John Miller wrote:
Let's say person A is using SIP provider A on an ATA, and person B is using
SIP provider B on another ATA.

Person A calls person B (both using SIP, but with "real" phone numbers, so
not via ENUM). Which devices are deciding (comparing their codec priority
list) which codec to use?

- Provider A and provider B? If this is the case, can it be that the link
between provider B and the ATA of person B can use another codec? Also for
the link of person A's ATA to provider A?

- Provider A and person's B's ATA? Is this case provider B is
"transparant"?

- Person A's ATA and person B's ATA? In this case both provider's priority
list doesn't matter?


There are a number of answers to this question depending on how provider
A and B have configured their network and how they interconnect.

As a general rule whatever you are directly conencting to i.e UA, B2BUA,
media relay will negoitate a codec that is common to both you and them.
It is possible that the interconnect, if used, will use a totally
different protocol and will be performing transcoding to enable this.

HTH
Regards

Jon
  #5  
Old May 27th 07, 12:58 PM posted to uk.telecom.voip
John Miller
external usenet poster
 
Posts: 15
Default Who decides which codec to use?

b) Provider A and Provider B do not "know" each other: Provider A
will built a PSTN connection to Provider B's network.
In this case, there is no link between the codes used.


Isn't G.711 (a in Europe, and in America) the international standard for
PSTN calls? I just thought so because ISDN is also using this codec, and I
suppose the PSTN operators are avoiding extra conversions on the way.

So I thought if my ATA would negociate G.711a to my SIP provider, that there
would be less conversions needed?


  #6  
Old May 27th 07, 01:05 PM posted to uk.telecom.voip
John Miller
external usenet poster
 
Posts: 15
Default Who decides which codec to use?

As a general rule whatever you are directly conencting to i.e UA, B2BUA,
media relay will negoitate a codec that is common to both you and them. It
is possible that the interconnect, if used, will use a totally different
protocol and will be performing transcoding to enable this.


Don't all these codec conversions introduce a degrade in quality?

For instance, some people use a DECT phone which is using G.726. This is
converted to analogue to their ATA. This ATA is using maybe G.729 to the
VOIP provider. This VOIP provider is putting this signal onto the PSTN via
G.711a (I assume). And at the other side a similar conversion could be
done...

It just seems like a quality loss is unavoidable this way.


  #7  
Old May 27th 07, 04:49 PM posted to uk.telecom.voip
Jon Farmer
external usenet poster
 
Posts: 40
Default Who decides which codec to use?

John Miller wrote:
As a general rule whatever you are directly conencting to i.e UA, B2BUA,
media relay will negoitate a codec that is common to both you and them. It
is possible that the interconnect, if used, will use a totally different
protocol and will be performing transcoding to enable this.


Don't all these codec conversions introduce a degrade in quality?

For instance, some people use a DECT phone which is using G.726. This is
converted to analogue to their ATA. This ATA is using maybe G.729 to the
VOIP provider. This VOIP provider is putting this signal onto the PSTN via
G.711a (I assume). And at the other side a similar conversion could be
done...

It just seems like a quality loss is unavoidable this way.



Yes you are correct, the quality of the call will be, amongst other
things, governed by the codec which uses the least amount of
bandwidth/least efficient conversion algorithm.

An analogy would be the internet itself where by the speed from your PC
to a server for example is governed by the slowest link in the route
from you to the server.

HTH

Regards
Jon
  #8  
Old May 29th 07, 11:45 PM posted to uk.telecom.voip
alexd
external usenet poster
 
Posts: 1,765
Default Who decides which codec to use?

John Miller wrote:

Isn't G.711 (a in Europe, and in America) the international standard for
PSTN calls? I just thought so because ISDN is also using this codec, and
I suppose the PSTN operators are avoiding extra conversions on the way.

So I thought if my ATA would negociate G.711a to my SIP provider, that
there would be less conversions needed?


Potentially - but how do you know how your SIP provider gets your call
to/from the PSTN? They may be using ISDN, but then again they may have an
IP trunk instead.

--
http://ale.cx/ (AIM:troffasky) )
22:44:15 up 31 days, 45 min, 2 users, load average: 0.05, 0.23, 0.20
09 f9 11 02 9d 74 e3 5b d8 41 56 c5 63 56 88 c0

  #9  
Old May 30th 07, 10:37 AM posted to uk.telecom.voip
Gordon Henderson
external usenet poster
 
Posts: 64
Default Who decides which codec to use?

In article ,
John Miller wrote:
Let's say person A is using SIP provider A on an ATA, and person B is using
SIP provider B on another ATA.

Person A calls person B (both using SIP, but with "real" phone numbers, so
not via ENUM). Which devices are deciding (comparing their codec priority
list) which codec to use?

- Provider A and provider B? If this is the case, can it be that the link
between provider B and the ATA of person B can use another codec? Also for
the link of person A's ATA to provider A?

- Provider A and person's B's ATA? Is this case provider B is
"transparant"?

- Person A's ATA and person B's ATA? In this case both provider's priority
list doesn't matter?


coming into this a bit late, but what you'll most likely find is that the
Providers won't let you use any CODEC other than G711 (u or a) and the
reason for that is likely to be down to the CPU horsepower required to
do the transcoding. pA and pB are going to be using the PSTN to talk to
each other (unless they have private peering arrangements) and, as been
pointed out that will almost certainly want to use G711.

I only allow my subscribers to connect in on G711 (a or u as the
conversion between the 2 is negligible), but I also allow G726 which
is half the bandwidth and the conversion CPU usage is very low, and
very occasionally, GSM, but only from PBXs I install at the clients end,
talking to one of my head-end switches, and only then if their Internet
line is anything other than "excellent".

Sipgate don't allow GSM, but I haven't tried others.

I have heard of some US providers using G729 which is patent encumbered,
(the license is $10 per line) but most phones/ata's support that anyway
- good quality and very minimal bandwidth, but how they're handling the
compute required to transcode it to G711 which they (probably) need to
do to connect to the PSTN is anyones guess. (Although it might well be
that the expense of additional CPU/Servers is cheaper than the additional
bandwidth required when dealing with 1000's or more subscribers)

With regard to the original question of "who decides", there is a
negotiation phase during call setup, each end has a list of CODECs they
support and usually the highest quality one will 'win', but it's almost
always going to be the upstream provider that has the final say, not the
end-user - and you can get into situations where you simply can't place
the call because the provider doesn't (or won't) support the CODECs the
end-user has available.

For an example of the CPU time required to do some translations, have a
look at this:

Translation times between formats (in milliseconds)
Source Format (Rows) Destination Format(Columns)

g723 gsm ulaw alaw g726 adpcm slin lpc10 g729 speex ilbc
g723 - - - - - - - - - - -
gsm - - 10 10 32 12 9 43 - 259 173
ulaw - 33 - 1 24 4 1 35 - 251 165
alaw - 33 1 - 24 4 1 35 - 251 165
g726 - 51 20 20 - 22 19 53 - 269 183
adpcm - 34 3 3 25 - 2 36 - 252 166
slin - 32 1 1 23 3 - 34 - 250 164
lpc10 - 57 26 26 48 28 25 - - 275 189
g729 - - - - - - - - - - -
speex - 56 25 25 47 27 24 58 - - 188
ilbc - 64 33 33 55 35 32 66 - 282 -

This is the output from an asterisk box command:
show translation recalc 30

The times are in ms, and as this is running on a 533MHz processor, they
do appear quite large, but it's a good demonstration and highlights the
overheads better than running on a 3GHz processor (where they may be
sub millisecond, but when you're handing 100s of calls, it all adds up!)

And you can also see that speex and ilbc, which, while nice geeky, high
compression CODECS, do require a lot of CPU to do the encoding and
decoding... (and neither of them sound particularly good IMO - unless you
like sounding like a Dalek

Gordon
  #10  
Old May 30th 07, 10:24 PM posted to uk.telecom.voip
Jon Farmer
external usenet poster
 
Posts: 40
Default Who decides which codec to use?

Gordon Henderson wrote:
In article ,
John Miller wrote:
Let's say person A is using SIP provider A on an ATA, and person B is using
SIP provider B on another ATA.

Person A calls person B (both using SIP, but with "real" phone numbers, so
not via ENUM). Which devices are deciding (comparing their codec priority
list) which codec to use?

- Provider A and provider B? If this is the case, can it be that the link
between provider B and the ATA of person B can use another codec? Also for
the link of person A's ATA to provider A?

- Provider A and person's B's ATA? Is this case provider B is
"transparant"?

- Person A's ATA and person B's ATA? In this case both provider's priority
list doesn't matter?


coming into this a bit late, but what you'll most likely find is that the
Providers won't let you use any CODEC other than G711 (u or a) and the
reason for that is likely to be down to the CPU horsepower required to
do the transcoding. pA and pB are going to be using the PSTN to talk to
each other (unless they have private peering arrangements) and, as been
pointed out that will almost certainly want to use G711.


I allow

G711u/a
GSM
G726-32
G729

Regards

Jon


 




Currently Active Users Viewing This Thread: 1 (0 members and 1 guests)
 
Thread Tools
Display Modes

Posting Rules
You may not post new threads
You may not post replies
You may not post attachments
You may not edit your posts

vB code is On
Smilies are On
[IMG] code is On
HTML code is Off
Forum Jump

Similar Threads
Thread Thread Starter Forum Replies Last Post
Best codec for using a modem on Simon Finnigan uk.telecom.voip (UK VOIP) 18 August 12th 06 02:05 PM
codec > uk.telecom.voip (UK VOIP) 2 March 6th 06 10:25 AM
Which Codec? the hamiltons uk.telecom.voip (UK VOIP) 1 January 21st 06 02:41 PM
preferred codec? Lee uk.telecom.voip (UK VOIP) 2 July 27th 05 05:20 PM
Which codec corresponds to GSM? Henry F III uk.telecom.voip (UK VOIP) 0 July 14th 05 10:18 AM


All times are GMT +1. The time now is 11:21 AM.


Powered by vBulletin® Version 3.6.4
Copyright ©2000 - 2019, Jelsoft Enterprises Ltd.Content Relevant URLs by vBSEO 2.4.0
Copyright 2004-2019 BroadbanterBanter.
The comments are property of their posters.