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.

Forcing inbound Codec to G711



 
 
Thread Tools Display Modes
  #1  
Old December 8th 06, 04:25 PM posted to uk.telecom.voip
Pet @ www.gymratz.co.uk )
external usenet poster
 
Posts: 53
Default Forcing inbound Codec to G711

At a guess I presume inbound numbers from public PSTN routed from ones
VOIP supplier would be routed on the most economical codec eg G712A/B

If that's the case, then if I were to force the voip ports to a single
codec e.g. G711MU, presumably the inbound call would be forced to
upscale to this single codec rather than the lowest it could get away with?

Would there be any forseeable disadvantages doing this?
I see it as a way of getting max. call quality in inbound calls esp.
where downstream bandwidth is not such an issue as upstream.

Any thoughts?
Pete
--
http://gymratz.co.uk - Best Gym Equipment & Bodybuilding Supplements UK.
http://gymratz.co.uk/polar-heart-rate-monitors/ Polar HeartRate Monitors
http://fitness-equipment-uk.com - UK's No.1 Fitness Equipment Suppliers.
http://water-rower.co.uk - Worlds best prices on the Worlds best Rower.
  #2  
Old December 9th 06, 10:11 AM posted to uk.telecom.voip
alexd
external usenet poster
 
Posts: 331
Default Forcing inbound Codec to G711

"Pet @ www.gymratz.co.uk ;)" wrote:

At a guess I presume inbound numbers from public PSTN routed from ones
VOIP supplier would be routed on the most economical codec eg G712A/B


What do you mean by "economical"? Some codecs cost less bandwidth but more
CPU time. Some codecs are patented [whether legally or not] and require
licensing.

If that's the case, then if I were to force the voip ports to a single
codec e.g. G711MU, presumably the inbound call would be forced to
upscale to this single codec rather than the lowest it could get away
with?


If it's SIP we're talking about, both ends have to agree on a codec. If they
cannot, the call will fail.

Would there be any forseeable disadvantages doing this?


Higher bandwidth use.

I see it as a way of getting max. call quality in inbound calls esp.
where downstream bandwidth is not such an issue as upstream.


When budgeting bandwidth, it makes sense to assume that it will be
symmetrical. When I watch the bandwidth graph on my Asterisk server [SIP,
G711u], it's always level in both directions, ie it doesn't have peaks and
troughs that correlate with speech and silence. This of course may depend
on silence suppression and voice activity detection settings etc on the
endpoints.

--
http://ale.cx/ (AIM:troffasky) )
09:58:29 up 19 days, 13:41, 2 users, load average: 0.02, 0.06, 0.02
This is my BOOOOOOOOOOOOOOOOOOOOOMSTICK

  #3  
Old December 9th 06, 11:45 AM posted to uk.telecom.voip
Philippe Deleye
external usenet poster
 
Posts: 76
Default Forcing inbound Codec to G711


"Pet @ www.gymratz.co.uk ;)" wrote in message
. uk...
If that's the case, then if I were to force the voip ports to a single
codec e.g. G711MU, presumably the inbound call would be forced to
upscale to this single codec rather than the lowest it could get away

with?

If the other end of the call (your Providers PSTN Gateway) does not support
the codec you try to enforce, then the call will fail.

I see it as a way of getting max. call quality in inbound calls esp.
where downstream bandwidth is not such an issue as upstream.


There is no connection between inbound calls and downstream bandwith. Both
Inbound calls and outbound are using the same (symmetric) bandwith, if using
the same codec.
With good downstraem bandwith, you will experience good sound quality, but
with a bad upstraeam bandwith the other party will experience bad quality
(what you hear is downstream, what the other party hears is upstream - not
related to incoming our outgoing calls ...


  #4  
Old December 10th 06, 11:59 AM posted to uk.telecom.voip
Jono
external usenet poster
 
Posts: 1,539
Default Forcing inbound Codec to G711

alexd brought next idea :


When budgeting bandwidth, it makes sense to assume that it will be
symmetrical. When I watch the bandwidth graph on my Asterisk server [SIP,
G711u],


How'd you do that?


  #5  
Old December 11th 06, 09:54 AM posted to uk.telecom.voip
Paul
external usenet poster
 
Posts: 36
Default Forcing inbound Codec to G711

Pet @ www.gymratz.co.uk ;) wrote:
At a guess I presume inbound numbers from public PSTN routed from ones
VOIP supplier would be routed on the most economical codec eg G712A/B


Not in my experience, most stick to G.711a/u. It's still only needs
about 80kbps per call which is fine for the vast majority of Internet
connections these days. G729 would save on bandwidth but it's CPU
intensive, this might not matter for you but it'll make a big difference
for a service provider. It is also not a free codec and the license
holder must be paid for it. I dare say that most importantly, it's
lower quality than G711 (and therefore the PSTN) so the customer would
perceive the service provider's service as poorer quality.


If that's the case, then if I were to force the voip ports to a single
codec e.g. G711MU, presumably the inbound call would be forced to
upscale to this single codec rather than the lowest it could get away with?


Yes if you set your VoIP hardware to only accept g711a/u then either the
service providers server will agree to this or the call will fail.


Would there be any forseeable disadvantages doing this?
I see it as a way of getting max. call quality in inbound calls esp.
where downstream bandwidth is not such an issue as upstream.


Less calls through your Internet connection. The calls require the same
amount of bandwidth in either direction regardless of whether they are
incoming calls or outgoing calls.


Any thoughts?
Pete

  #6  
Old December 11th 06, 12:51 PM posted to uk.telecom.voip
ale.cx
external usenet poster
 
Posts: 63
Default Forcing inbound Codec to G711

On Dec 10, 11:59 am, Jono wrote:

How'd you do that?


gkrellmd on the Asterisk box, gkrellm on my desktop :-D

alexd

  #7  
Old December 11th 06, 02:12 PM posted to uk.telecom.voip
Pet @ www.gymratz.co.uk )
external usenet poster
 
Posts: 53
Default Forcing inbound Codec to G711 & SIP port numbers

Paul wrote:

Not in my experience, most stick to G.711a/u. It's still only needs
about 80kbps per call which is fine for the vast majority of Internet
connections these days. G729 would save on bandwidth but it's CPU
intensive, this might not matter for you but it'll make a big difference
for a service provider. It is also not a free codec and the license
holder must be paid for it. I dare say that most importantly, it's
lower quality than G711 (and therefore the PSTN) so the customer would
perceive the service provider's service as poorer quality.


I was just going down the train of thought based on last weeks incoming
calls, though, they were being routed through a sipgate number when our
BT line was busy. Since then I have signed up to voipfone.co.uk mainly
for incoming calls (on BT busy) but also for the CLI on outgoing calls
should we have anyone that decides not to answer "no ID/International"
calls from voipcheap.com

snip

Less calls through your Internet connection. The calls require the same
amount of bandwidth in either direction regardless of whether they are
incoming calls or outgoing calls.


I'll leave it with multiple codecs available then and see how we go.

On a further note, if I have 2 outgoing lines via router do I need to
specify a range of SIP ports
e.g. 5060-5061 for line 1 and 5062 - 5063 for line 2

At the moment all accounts are left at 5060.

Cheers
Pete


--
http://gymratz.co.uk - Best Gym Equipment & Bodybuilding Supplements UK.
http://gymratz.co.uk/polar-heart-rate-monitors/ Polar HeartRate Monitors
http://fitness-equipment-uk.com - UK's No.1 Fitness Equipment Suppliers.
http://water-rower.co.uk - Worlds best prices on the Worlds best Rower.
  #8  
Old December 12th 06, 05:13 PM posted to uk.telecom.voip
ale.cx
external usenet poster
 
Posts: 63
Default Forcing inbound Codec to G711 & SIP port numbers

On Dec 11, 2:12 pm, "Pet @ www.gymratz.co.uk ;)"
wrote:

On a further note, if I have 2 outgoing lines via router do I need to
specify a range of SIP ports
e.g. 5060-5061 for line 1 and 5062 - 5063 for line 2


For outgoing you don't need to forward any ports so long as your router
does NAT and the firewall rules don't block it.

For incoming it's slightly different. If it's one physical device [eg a
single router, ATA or Asterisk server] with multiple accounts on, the
device should be able to deduce from the info in the inbound call which
account the call is for. SIP calls are a bit like email addresses, the
bit before the '@' is the account that the call is for.

If you've got multiple devices registering their own accounts behind a
single public IP address, you will need to make sure the relevant ports
are forwarded to the relevant devices for them to be able to receive
calls.

At the moment all accounts are left at 5060.


Does it work?

  #9  
Old December 12th 06, 05:40 PM posted to uk.telecom.voip
Pet @ www.gymratz.co.uk )
external usenet poster
 
Posts: 53
Default Forcing inbound Codec to G711 & SIP port numbers

ale.cx wrote:

For outgoing you don't need to forward any ports so long as your router
does NAT and the firewall rules don't block it.


That'd be ok then.
It's only multiple outgoing calls that concerned me.

For incoming it's slightly different. If it's one physical device [eg a
single router, ATA or Asterisk server] with multiple accounts on, the
device should be able to deduce from the info in the inbound call which
account the call is for. SIP calls are a bit like email addresses, the
bit before the '@' is the account that the call is for.

If you've got multiple devices registering their own accounts behind a
single public IP address, you will need to make sure the relevant ports
are forwarded to the relevant devices for them to be able to receive
calls.


At the moment all accounts are left at 5060.


Does it work?


I haven't really tried it in anger.
Currently incoming BT calls are set to divert to 0845 number provided by
voipfone should the bt line be busy. from there both voip ports on the
router ring so should bt line be tied up and port 1 also tied up then
port 2 should get the call.

We don't generally get enough calls to try it and most of the time
there's only 2 of us here anyway.
:)

It just seemed that when I was trying to make 2 outgoing voip calls at
the same time I was getting a busy tone, hence the reason for the post.

May just have been a coincidence though.

Cheers
Pete


--
http://gymratz.co.uk - Best Gym Equipment & Bodybuilding Supplements UK.
http://gymratz.co.uk/polar-heart-rate-monitors/ Polar HeartRate Monitors
http://fitness-equipment-uk.com - UK's No.1 Fitness Equipment Suppliers.
http://water-rower.co.uk - Worlds best prices on the Worlds best Rower.
  #10  
Old December 18th 06, 01:23 PM posted to uk.telecom.voip
PeterW
external usenet poster
 
Posts: 49
Default Forcing inbound Codec to G711

Paul wrote in
:

Pet @ www.gymratz.co.uk ;) wrote:
At a guess I presume inbound numbers from public PSTN routed from
ones VOIP supplier would be routed on the most economical codec eg
G712A/B


Not in my experience, most stick to G.711a/u. It's still only needs
about 80kbps per call which is fine for the vast majority of Internet
connections these days. G729 would save on bandwidth but it's CPU
intensive, this might not matter for you but it'll make a big
difference for a service provider. It is also not a free codec and
the license holder must be paid for it. I dare say that most
importantly, it's lower quality than G711 (and therefore the PSTN) so
the customer would perceive the service provider's service as poorer
quality.


If that's the case, then if I were to force the voip ports to a
single codec e.g. G711MU, presumably the inbound call would be forced
to upscale to this single codec rather than the lowest it could get
away with?


Yes if you set your VoIP hardware to only accept g711a/u then either
the service providers server will agree to this or the call will fail.


Would there be any forseeable disadvantages doing this?
I see it as a way of getting max. call quality in inbound calls esp.
where downstream bandwidth is not such an issue as upstream.


Less calls through your Internet connection. The calls require the
same amount of bandwidth in either direction regardless of whether
they are incoming calls or outgoing calls.


Any thoughts?
Pete



Most offer G.711a (this is the European/worldwide standard). It provides
the best 8khz sampled audio available.

G.711u is a US standard. G.726/729 or other compression standards (GSM,
Speex etc.) is only useful where bandwidth is a big issue.

peter
 




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 01:05 PM
codec > uk.telecom.voip (UK VOIP) 2 March 6th 06 09:25 AM
Which Codec? the hamiltons uk.telecom.voip (UK VOIP) 1 January 21st 06 01:41 PM
preferred codec? Lee uk.telecom.voip (UK VOIP) 2 July 27th 05 04:20 PM
Which codec corresponds to GSM? Henry F III uk.telecom.voip (UK VOIP) 0 July 14th 05 09:18 AM


All times are GMT +1. The time now is 12:33 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.