there are so many factors in your setting that can affect your bandwith. for example if you are using G723r63 for one channel without AVD and I frame per packet this is what you will consume:
 
Codec payload = 6.3K
codec payload + RTP =9.6K
Codec payload + RTP + UDP = 11K
codec payload + RTP +UDP + IP =  17K.  Total per channel.
 
if you set your codec to 2 frames per packet the total bandwith will go down to 11K from 17K.  and when you add AVD then bandwith can go down to  an avarage of 4K per channell but you still need the maximum bandwidth 11K or 17K per channel depending on the number frames per packet. the advantage of AVD is that you will be able to sequeeze in more channels oan the same bandwidth.
 
thanks
 
Mohamed omar
 
 
Nyamul Hassaan <nyamul@gmail.com> wrote:
Hi All,

Mohammed, which IP Phones are you using?

There is something also I'm confused about all this Codec and bandwidth issue.
I've seen that when I'm talking using my computer via a softphone, G.723 and G.729 takes approximately 24kbps and 32kbps respectively of bandwidth.
But, in my terminating dedicated boxes (Quintum in my case), I see that I need to allocate only 14-15 kbps and 17-18 kbps respectively.  Sometimes, total bandwidth is even less.
I've a 768kbps connection from my ISP under which I'm running around 50 lines.  Still, the consumption doesn't go over 600kbps even during peak hours. All my customers are using G.723r63.
Any ideas on why this is happening?

Regards
HASSAAN



On 5/27/05, Mohamed Omar <amatek2004@yahoo.ca> wrote:
I use dialup connection with my IPPhones which has a buildin modem without any issue on both codec G723 and G729.  I sometime get better quality on the some phone if I use dialup then when I was braodband.  this happens when I call most aftrican country where the termination is not good.   This is because when I use broadband the media has to through some kinds of NAT transval while the dialup normally gets public IP so no NAT transval and the media gets the shorts path.
 
thanks
 
Mohamed
 


Jorge Crichigno <jcrichigno@conexion.com.py> wrote:
I think the data rates with overhead are as follow:

----------------------------------------------------------------
iLBC - 30 ms (a packet each 30 ms)

RTP payload: 50 bytes --> Rate: 13.3 Kbps
RTP: 62 bytes
UDP: 70 bytes
IP: 90 bytes --> total rate: 24 Kbps

-------------------------------------------------------------------
iLBC - 20 ms (a packet each 20 ms)

RTP payload: 38 bytes --> Rate: 15.2 Kbps
RTP: 50 bytes
UDP: 58 bytes
IP: 78 bytes --> total rate: 31.2 Kbps

-------------------------------------------------------------------
G.729 - 2 voice frame per packet

RTP payload: 20 bytes --> Rate: 8 Kbps
RTP: 32 bytes
UDP: 40 bytes
IP: 60 bytes --> Total rate: 24 Kbps

-------------------------------------------------------------------
G.729 - 4 voice frame per packet

RTP payload: 40 bytes --> Rate: 8 Kbps
RTP: 52 bytes
UDP: 60 bytes
IP: 80 bytes --> Total rate: 16 Kbps

-------------------------------------------------------------------
G.711 - 20 ms (a packet each 20 ms)

RTP payload: 160 bytes --> Rate: 64 Kbps
RTP: 172 bytes
UDP: 180 bytes
IP: 200 bytes --> Total rate: 80 Kbps



El vie, 27-05-2005 a las 10:36, Iqbal escribi?:
> the 5.3, 6.3K are really theoretical, i dont think they include IP
> overheads, I used media_sessions.phtml, and looked at the actual calls
> per codec, and I dont think u can really get a good call without 50-70K,
> also most bandwidth providers (at least here in the UK) are
> asymmetrical, so even on a 128K, u could have problems.
>
> Having said that I have done a nice call on xlite using ilbc on dial up.
>
> Iqbal
>
> Kofi Obiri-Yeboah wrote:
>
> >Just to add a fe w more details, Greger is right to point out the quality
> >inferiority of G.723 compared to those of G711 and G729. In fact, in most VOIP
> >deployments, in order to quarantee interoperability, a minimum bandwidth of
> >128K is specified. However to reach the wider "lower bandwidth areas" most
> >service providers are opting for G.723 which uses either 5.3 or 6.3K. At this
> >low bandwidth transmission needs, one could literally reach "dial up modem"
> >equipped areas. in fact most VOIP phone hardware and software are begining to
> >specify G.723 as their default codec. Note that until the direct media
> >connection phase of a VOIP vall setup, wide bandwidth is not required. Also
> >note that analogue phones have a maximim bandwidth need of 3K, hence even the
> >low quality of G.723/5.3K, compared to the average analogue phone call, is
> >superior
> >
> >
> < BR>> _______________________________________________
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