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Sat 17 of May, 2008 [12:56 UTC]

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  • Juan Ortega, Thu 15 of May, 2008 [10:33 UTC]: Hi everybody, I'm Juan, an ITCom student, and I need to know what basic elements I need to create a VoIP network. Can anybody helpme, please?,Thank you very much
  • gineta, Wed 14 of May, 2008 [03:58 UTC]: any here not fine the configuration of firewall juniper -screem for VOIP asterisk????
  • Anoop Prabhakaran, Tue 13 of May, 2008 [12:16 UTC]: I am developing Asterisk IVR, Whenever i make a internation call to the IVR system, the DTMF is not getting detected properly, this happens only for the first time, second call onwards system works fine. why this is happening
  • joe, Mon 12 of May, 2008 [04:27 UTC]: Is there an opensource browser based softphone, or a system like Busta where everything is not manages through their website?
  • Nick Barnes, Fri 09 of May, 2008 [11:36 UTC]: Christopher - yesterday I tried an Asterisk install on a CentOS 5.1 box with stock GUI and it all worked fine. Sorry I can't help.
  • aero, Fri 09 of May, 2008 [08:20 UTC]: can someone help me out on this, i tried to play some sound files on my asterisk box and this is the error message i got. WARNING[4429]: format_wav.c:169 check_header: Unexpected freqency 22050 May 8 11:17:39 WARNING[4433]: codec_gsm.c:194 gsmtolin_fra
  • Christopher Faust, Thu 08 of May, 2008 [14:15 UTC]: I beleive that I may have to change something in the xserver configuration. Please advise
  • Christopher Faust, Thu 08 of May, 2008 [14:14 UTC]: Everything was perfect. In the bios I have increased the memory allocated Still receive input not supported on my display.
  • Christopher Faust, Thu 08 of May, 2008 [14:13 UTC]: This would not be my main box. I am doing some testing to see if I can install zaptel and asterisk 1.4 on a full centos 5.1 box with development software Its bizzare, because before I went through the asterisk and zaptel installation everything was perfe
  • Nick Barnes, Thu 08 of May, 2008 [13:44 UTC]: Christopher - I can't see any way in which an Asterisk installation would muck your GUI, but remember that it is advised not to use a GUI on an Asterisk box anyway.
Server Stats
  • Execution time: 0.42s
  • Memory usage: 2.21MB
  • Database queries: 29
  • GZIP: Disabled
  • Server load: 1.04

Bandwidth consumption

VOIP Bandwidth consumption naturally depends on the codec used.

When calculating bandwidth, one can't assume that every channel is used all the time. Normal conversation includes a lot of silence, which often means no packets are sent at all. So even if one voice call sets up two 64 Kbit RTP streams over UDP over IP over Ethernet (which adds overhead), the full bandwidth is not used at all times.

A codec that sends a 64kb stream results in a much larger IP network stream. The main cause of the extra bandwidth usage is IP and UDP headers. VoIP sends small packets and so, many times, the headers are actually much larger than the data part of the packet.

IAX2 trunking helps with the IP overhead, but only when you are sending more than 2 or so calls between the same Asterisk servers. John Todd has done some useful practical testing, named IAX2 trunking: codec bandwidth comparison notes and results.

The bandwidth used depends also on the datalink (layer2) protocols. Several things influence the bandwidth used, payload size, ATM cell headers, VPN headers, use of header compression and IAX2 Trunked. You can see the influence of some of this factors using the Asteriskguide bandwidth calculator.

http://www.terracall.com/FAQs_white_1.aspx has this table which shows how the codec's theoretical bandwidth usage expands with UDP/IP headers:

 Codec       BR       NEB
 G.711     64 Kbps  87.2 Kbps
 G.729      8 Kbps  31.2 Kbps
 G.723.1  6.4 Kbps  21.9 Kbps
 G.723.1  5.3 Kbps  20.8 Kbps
 G.726     32 Kbps  55.2 Kbps
 G.726     24 Kbps  47.2 Kbps
 G.728     16 Kbps  31.5 Kbps
 iLBC      15 Kbps  27.7 Kbps
 
BR = Bit rate
NEB = Nominal Ethernet Bandwidth (one direction)


Created by oej, Last modification by Evert Meulie on Tue 04 of Dec, 2007 [10:12 UTC]

Comments Filter

New Release for VoIP Blocking solutions

by jenniferhan on Friday 11 of January, 2008 [09:16:47 UTC]
We have realeased our VoIP anti Blocking solutions which include:

VG Plugin software for Windows operation system,
VGSC software for Windows Operation system,
VGBC hardware for terminal side
VG SPE Gateways for terminal side.

Please contact me for more information. Thank you.

Andy Wong
Email: Xd.wong@speed-voip.com
MSN: andywong-01@hotmail.com

VPN for VoIP Blocking

by jenniferhan on Wednesday 12 of December, 2007 [03:39:03 UTC]
Somebody use VPN to solve the VoIP Blocking issue. But it seems not a good way to solve the voip blocking issue. Because VPN will take more bandwidth and will take effection on the Voice Quality

Currently I am using the VGCP, a new solution to solve the VoIP Blocking issue. Following is theirs website:
http://www.speed-voip.com/index-36.html

If any of you have interested, you may try to use it to solve your VoIP Blocking problems. Thanks.

Andy
andywong-01@hotmail.com

VoIP Bandwidth Calaculation

by daveg1k on Thursday 10 of June, 2004 [08:59:00 UTC]
For a White Paper on how to calculate VoIP bandwidth on Ethernet see: VoIP Bandwidth Calculation White Paper
Also see the on-line VoIP Bandwidth Calaculator
Edit

test

by Anonymous on Friday 02 of January, 2004 [07:40:45 UTC]
test

Another link

by blackfire on Wednesday 03 of December, 2003 [15:09:07 UTC]
Here's a link to John Todd's analysis of bandwidth usage with IAX2.
http://lists.digium.com/pipermail/asterisk-users/2003-June/014788.html
Also, AFAIK, IAX2 trunking is used above 10 calls (due to 40-byte overhead vs 4-byte on single calls)

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