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.

[email protected], conferences, codecs and bandwidth



 
 
Thread Tools Display Modes
  #1  
Old May 21st 06, 04:43 PM posted to uk.telecom.voip
linker3000
external usenet poster
 
Posts: 75
Default [email protected], conferences, codecs and bandwidth

Hi guys,

I posted this in the [email protected] forums but hope I might catch a few more
helpful people here too...

I'd appreciate some real-world experience if possible:

If we roll out our * trial, we will initially have about 30 IP phones
(no analogues), one per branch office, and I was wondering about
conference/bandwidth issues...

The server is a dual core P4-2.6GHz with 1GiB RAM so I guess we have
enough steam to start with?! At the moment the server is conncted to a
2Mbit ADSL line with an upstream speed of 288Kb. I will probably upgrade
this to 8Mbit/448Kb service.

Here's the main questions:

How's the system likely to cope with all 30 sites in a conference over
the current/upgraded ADSL line?

I understand that the choice/mix of codecs used when conferencing can
have a big impact on the results - our current phone of choice supports
the usual a-law and u-law stuff and others will also do ilbc and gsm -
obviously, the latter two use less bandwidth but what is likely to
happen if we mix them with phones on a-law/u-law or should we just stick
to one set of common codecs - which would be the bigger bandwidth
a/u-law ones?

I have seen some dev threads + others on silence suppression and keeping
it OFF in order to provide the proper timing for RTP streams - does this
still apply with [email protected] 2.8?

How does the ADSL bandwidth relate to max number of simultaneous
connections? I can appreciate a simple calculation of (available
bandwidth / codec bandwidth requirement), but guess in the real world
it's not as simple as that when considering things such as simultaneous
use of channels not actually transmitting/receiving 100% of the time,
but how aboout the streams associated with conferences and also the
'overhead' of transmitting all the silence?

I have had a good look around the main Asterisk and [email protected] sites
and have found some generalised info on all of this, but not a lot of
real world input.

Any insight appreciated.

Thanks
  #2  
Old May 21st 06, 08:34 PM posted to uk.telecom.voip
stephen
external usenet poster
 
Posts: 381
Default [email protected], conferences, codecs and bandwidth

"linker3000" wrote in message
...
Hi guys,

I posted this in the [email protected] forums but hope I might catch a few more
helpful people here too...

I'd appreciate some real-world experience if possible:

If we roll out our * trial, we will initially have about 30 IP phones
(no analogues), one per branch office, and I was wondering about
conference/bandwidth issues...

The server is a dual core P4-2.6GHz with 1GiB RAM so I guess we have
enough steam to start with?! At the moment the server is conncted to a
2Mbit ADSL line with an upstream speed of 288Kb. I will probably upgrade
this to 8Mbit/448Kb service.


i dont know the asterisk specifics - but your link is going to limit the no.
of simultaneous calls.

depending on the codec you use, you are going to need 80 Kbps or more for
G.711, or 24 Kbps for G.729.

you cant have header compression on the link (since the ISP routers wont be
set up for it), although you may be able to RTP compression.

if you are using an IPsec internet VPN as well, then that will add a fair
bit of padding.

Here's the main questions:

How's the system likely to cope with all 30 sites in a conference over
the current/upgraded ADSL line?


ignoring QoS and background traffic - 10 calls on the current link.

In reality, if there is any other traffic you should derate this - the"rule
of thumb" i use on Cisco call manager setup is based on cisco
recommendations.
it is 1/3 of the bandwidth for all the real time (in a corporate general
purpose net) - this assumes you are using QoS and you have other traffic on
the same links.

I understand that the choice/mix of codecs used when conferencing can
have a big impact on the results - our current phone of choice supports
the usual a-law and u-law stuff and others will also do ilbc and gsm -
obviously, the latter two use less bandwidth but what is likely to
happen if we mix them with phones on a-law/u-law or should we just stick
to one set of common codecs - which would be the bigger bandwidth
a/u-law ones?


a lot depends on what you want - but i much prefer G.711 conferencing.

I have seen some dev threads + others on silence suppression and keeping
it OFF in order to provide the proper timing for RTP streams - does this
still apply with [email protected] 2.8?


Not sure - but does it help?

silence suppression is based on the assumption that each line is "off" 50%
of the time in each direction for typical conversations.
But with a peer to peer conference the outbound link carries a "mix" of all
input - so it is going to be on most of the time. And outbound bandwidth is
where your bottleneck is.

How does the ADSL bandwidth relate to max number of simultaneous
connections? I can appreciate a simple calculation of (available
bandwidth / codec bandwidth requirement), but guess in the real world
it's not as simple as that when considering things such as simultaneous
use of channels not actually transmitting/receiving 100% of the time,
but how aboout the streams associated with conferences and also the
'overhead' of transmitting all the silence?

I have had a good look around the main Asterisk and [email protected] sites
and have found some generalised info on all of this, but not a lot of
real world input.

Any insight appreciated.


a lot depends on how your network is glued together and the implications
that has for actual useful VoIP bandwidth (and all quality issues such as
latency and loss rates)

if you expect to run 30 user conferences, then more bandwidth outbound. You
either need a better link or something symmetric such as SDSL.

Maybe 1 way to cut down the cost of bandwidth is to have the server hosted
somewhere? you usually get 10M or more then.

Thanks

--
Regards

- replace xyz with ntl


  #3  
Old May 21st 06, 08:42 PM posted to uk.telecom.voip
linker3000
external usenet poster
 
Posts: 75
Default [email protected], conferences, codecs and bandwidth

stephen wrote:
"linker3000" wrote in message
...
Hi guys,

I posted this in the [email protected] forums but hope I might catch a few more
helpful people here too...

I'd appreciate some real-world experience if possible:

If we roll out our * trial, we will initially have about 30 IP phones
(no analogues), one per branch office, and I was wondering about
conference/bandwidth issues...

The server is a dual core P4-2.6GHz with 1GiB RAM so I guess we have
enough steam to start with?! At the moment the server is conncted to a
2Mbit ADSL line with an upstream speed of 288Kb. I will probably upgrade
this to 8Mbit/448Kb service.


i dont know the asterisk specifics - but your link is going to limit the no.
of simultaneous calls.

depending on the codec you use, you are going to need 80 Kbps or more for
G.711, or 24 Kbps for G.729.

you cant have header compression on the link (since the ISP routers wont be
set up for it), although you may be able to RTP compression.

if you are using an IPsec internet VPN as well, then that will add a fair
bit of padding.
Here's the main questions:

How's the system likely to cope with all 30 sites in a conference over
the current/upgraded ADSL line?


ignoring QoS and background traffic - 10 calls on the current link.

In reality, if there is any other traffic you should derate this - the"rule
of thumb" i use on Cisco call manager setup is based on cisco
recommendations.
it is 1/3 of the bandwidth for all the real time (in a corporate general
purpose net) - this assumes you are using QoS and you have other traffic on
the same links.
I understand that the choice/mix of codecs used when conferencing can
have a big impact on the results - our current phone of choice supports
the usual a-law and u-law stuff and others will also do ilbc and gsm -
obviously, the latter two use less bandwidth but what is likely to
happen if we mix them with phones on a-law/u-law or should we just stick
to one set of common codecs - which would be the bigger bandwidth
a/u-law ones?


a lot depends on what you want - but i much prefer G.711 conferencing.
I have seen some dev threads + others on silence suppression and keeping
it OFF in order to provide the proper timing for RTP streams - does this
still apply with [email protected] 2.8?


Not sure - but does it help?

silence suppression is based on the assumption that each line is "off" 50%
of the time in each direction for typical conversations.
But with a peer to peer conference the outbound link carries a "mix" of all
input - so it is going to be on most of the time. And outbound bandwidth is
where your bottleneck is.
How does the ADSL bandwidth relate to max number of simultaneous
connections? I can appreciate a simple calculation of (available
bandwidth / codec bandwidth requirement), but guess in the real world
it's not as simple as that when considering things such as simultaneous
use of channels not actually transmitting/receiving 100% of the time,
but how aboout the streams associated with conferences and also the
'overhead' of transmitting all the silence?

I have had a good look around the main Asterisk and [email protected] sites
and have found some generalised info on all of this, but not a lot of
real world input.

Any insight appreciated.


a lot depends on how your network is glued together and the implications
that has for actual useful VoIP bandwidth (and all quality issues such as
latency and loss rates)

if you expect to run 30 user conferences, then more bandwidth outbound. You
either need a better link or something symmetric such as SDSL.

Maybe 1 way to cut down the cost of bandwidth is to have the server hosted
somewhere? you usually get 10M or more then.
Thanks


Stephen,

Thanks - that kinda backs up my thoughts and I was actually scoping out
SDSL and hosted services this afternoon.

L3K
  #4  
Old May 24th 06, 04:36 PM posted to uk.telecom.voip
MartinC
external usenet poster
 
Posts: 3
Default [email protected], conferences, codecs and bandwidth

The half way house might be the Zen Office 8000 Max service for 79 ex
VAT.
With downstream speeds of up to 8Mbps and upstream speeds of up to
832Kbps.

Martin

 




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
[email protected] {{{{{Welcome}}}}} uk.telecom.voip (UK VOIP) 7 April 22nd 06 10:37 PM
Asterisk @ Home Jono uk.telecom.voip (UK VOIP) 6 September 10th 05 04:44 PM
Asterisk at Home Matt uk.telecom.voip (UK VOIP) 1 July 15th 05 10:59 PM
asterisk @ Home - Help Please? RH uk.telecom.voip (UK VOIP) 2 July 5th 05 06:28 PM
Home wireless LAN bandwidth throttling Anthony Bowles uk.telecom.broadband (UK broadband) 12 June 10th 04 06:07 PM


All times are GMT +1. The time now is 03:34 PM.


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.