2mbps Coming Out Next Month
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Bandwidth for VOIP (kbps) 55.20% of time you speak 40.00Actual download bandwidth 33.12# Hrs VOIP used everyday 1.00# Seconds 3,600.00# Seconds VOIP Used in a Month 108,000.00Data Downloaded (bytes) 3,576,960.00Data Downloaded (KB) 447,120.00Data Downloaded (MB) 436.64Calculations have been done with G.726 whose bit rate is 32 kbps. 55.2 bitrate is after compensating for IP overheads.
You can use G.729 (with 8 kbps bitrate) and bring down effective bitrate to 15.4 kbps. That would bring data download to 122 MB.
For sake of comparison, GSM cell phones uses a RPE-LTP codec with 13 kbps bitrate. That would result in ~160 MB download.
So, fiber backed POTS landline quality voice will cost ~440 MB, and Cell quality voice will cost ~ 160 MB. Onlice chat quality voice will cost ~ 122 MB.
coolbuddy_79
The Coolest One!!!
i use VoipStunt and it uses the G.711 codec, the softphone that is, and it's bitrate is 64 kbps...so around 30 MB per hr...so it gives me a limitation of only about 1 hr per day of VOIP calls... 
From http://www.skype.com/help/faq/technical.html
How much bandwidth does Skype use while I’m in a call?
Skype automatically selects the best codec depending on the connection between yourself and the person you are calling. On average, Skype uses between 3-16 kilobytes/sec depending on bandwidth available for other party, network conditions in between, callers CPU performance, etc.
How much bandwidth does Skype use when there are no active calls?
On average Skype uses 0-0.5 kilobytes/sec while idle. This is used mainly for contact presence updates. The exact bandwidth depends on many factors.
Which protocols does Skype use?
Skype uses a proprietary protocol which we have developed. We looked at many available protocols when designing Skype and none were good enough for us. We hope you agree![/b]
You can use these codecs with any SIP compliant service (Skype is not among them).
You can customize https://sip-communicator.dev.java.net/ to use any codec you wish. There are quite a few Open source SIP Server / Clients / Proxy are available. You can customize them to use any codec.
I use Asterisk (as proxy) to a few SIP services (Stanaphone and others) and they support G.xxxx codecs.
How much bandwidth does Skype use while I’m in a call?
Skype automatically selects the best codec depending on the connection between yourself and the person you are calling. On average, Skype uses between 3-16 kilobytes/sec depending on bandwidth available for other party, network conditions in between, callers CPU performance, etc.
How much bandwidth does Skype use when there are no active calls?
On average Skype uses 0-0.5 kilobytes/sec while idle. This is used mainly for contact presence updates. The exact bandwidth depends on many factors.
Which protocols does Skype use?
Skype uses a proprietary protocol which we have developed. We looked at many available protocols when designing Skype and none were good enough for us. We hope you agree![/b]
You can use these codecs with any SIP compliant service (Skype is not among them).
You can customize https://sip-communicator.dev.java.net/ to use any codec you wish. There are quite a few Open source SIP Server / Clients / Proxy are available. You can customize them to use any codec.
I use Asterisk (as proxy) to a few SIP services (Stanaphone and others) and they support G.xxxx codecs.
Never used that perticular client.
However, since Open Source clients are available for Linux/Windows/java, I usually compile something of my own.
Or you can install Asterisk (Not it works on Windows as well). Asterisk will serve as a proxy to your SIP server, and has decent support for codecs. Almost all clients will work fine with Asterisk. This way you can use pretty much any codec without changing your client.
On latency, we have to give up anyways.
Have a look at ping stats from my MTNL connection.
C:\>ping sphone.vopr.vonage.netPinging sphone.vopr.vonage.net [216.115.20.41] with 32 bytes of data:Reply from 216.115.20.41: bytes=32 time=217ms TTL=50Reply from 216.115.20.41: bytes=32 time=216ms TTL=50Reply from 216.115.20.41: bytes=32 time=218ms TTL=50Reply from 216.115.20.41: bytes=32 time=217ms TTL=50Ping statistics for 216.115.20.41: Packets: Sent = 4, Received = 4, Lost = 0 (0% loss),Approximate round trip times in milli-seconds: Minimum = 216ms, Maximum = 218ms, Average = 217msC:\>ping 192.168.55.1Pinging 192.168.55.1 with 32 bytes of data:Reply from 192.168.55.1: bytes=32 time=2ms TTL=64Reply from 192.168.55.1: bytes=32 time=2ms TTL=64Reply from 192.168.55.1: bytes=32 time=2ms TTL=64Reply from 192.168.55.1: bytes=32 time=2ms TTL=64Ping statistics for 192.168.55.1: Packets: Sent = 4, Received = 4, Lost = 0 (0% loss),Approximate round trip times in milli-seconds: Minimum = 2ms, Maximum = 2ms, Average = 2ms
So, WiFi adss less then 1% in terms of latency.
OTOH, this what IEEE specs say on VOIP QOS:
#Voice traffic should be marked to DSCP EF per the QoS Baseline and RFC 3246.
#Loss should be no more than 1 percent.
#One-way latency (mouth to ear) should be no more than 150 ms.[/b]
But overall, since default ping times are > 200 ms, you will never get POTS quality transmission irrespective of codec. If yo uswitch to a TCP based proxy instead of IP based, you can get POTS "sound" quality but that will have frustating delay due to TCP overhead.
However, since Open Source clients are available for Linux/Windows/java, I usually compile something of my own.
Or you can install Asterisk (Not it works on Windows as well). Asterisk will serve as a proxy to your SIP server, and has decent support for codecs. Almost all clients will work fine with Asterisk. This way you can use pretty much any codec without changing your client.
On latency, we have to give up anyways.
Have a look at ping stats from my MTNL connection.
C:\>ping sphone.vopr.vonage.netPinging sphone.vopr.vonage.net [216.115.20.41] with 32 bytes of data:Reply from 216.115.20.41: bytes=32 time=217ms TTL=50Reply from 216.115.20.41: bytes=32 time=216ms TTL=50Reply from 216.115.20.41: bytes=32 time=218ms TTL=50Reply from 216.115.20.41: bytes=32 time=217ms TTL=50Ping statistics for 216.115.20.41: Packets: Sent = 4, Received = 4, Lost = 0 (0% loss),Approximate round trip times in milli-seconds: Minimum = 216ms, Maximum = 218ms, Average = 217msC:\>ping 192.168.55.1Pinging 192.168.55.1 with 32 bytes of data:Reply from 192.168.55.1: bytes=32 time=2ms TTL=64Reply from 192.168.55.1: bytes=32 time=2ms TTL=64Reply from 192.168.55.1: bytes=32 time=2ms TTL=64Reply from 192.168.55.1: bytes=32 time=2ms TTL=64Ping statistics for 192.168.55.1: Packets: Sent = 4, Received = 4, Lost = 0 (0% loss),Approximate round trip times in milli-seconds: Minimum = 2ms, Maximum = 2ms, Average = 2ms
So, WiFi adss less then 1% in terms of latency.
OTOH, this what IEEE specs say on VOIP QOS:
#Voice traffic should be marked to DSCP EF per the QoS Baseline and RFC 3246.
#Loss should be no more than 1 percent.
#One-way latency (mouth to ear) should be no more than 150 ms.[/b]
But overall, since default ping times are > 200 ms, you will never get POTS quality transmission irrespective of codec. If yo uswitch to a TCP based proxy instead of IP based, you can get POTS "sound" quality but that will have frustating delay due to TCP overhead.
QUOTE(yogi @ Sep 27 2006, 12:40 PM) [snapback]63415[/snapback]
I dont think we've seen any higher speed on any isp which satisfies all parameters of what a connection should have ----
1. reliable = less than 1% downtime
2. dedicated speed as promised
3. unlimited data transfer
4. affordable (& not ridiculously priced, on par with the rest of the world)
[/b]
Agree with Vebmetal. Airtel meets all of those criteria.Plus good CC. Only problem is they should be avilable in
your area.
QUOTE(vebmetal @ Sep 28 2006, 12:18 PM) [snapback]63510[/snapback]
What would be killer is if people just opted for unlimited plans and didn't go for any of the lousy limited plans... then the ISPs would be forced to only offer unlimited packages, which is how it should be.
[/b]
So true. Broadband is meaningless with capping.
A limited broadband is only slightly betterthan dial up IMO.
I dont think we've seen any higher speed on any isp which satisfies all parameters of what a connection should have ----
1. reliable = less than 1% downtime
2. dedicated speed as promised
3. unlimited data transfer
4. affordable (& not ridiculously priced, on par with the rest of the world)
[/b]
Agree with Vebmetal. Airtel meets all of those criteria.Plus good CC. Only problem is they should be avilable in
your area.
QUOTE(vebmetal @ Sep 28 2006, 12:18 PM) [snapback]63510[/snapback]
What would be killer is if people just opted for unlimited plans and didn't go for any of the lousy limited plans... then the ISPs would be forced to only offer unlimited packages, which is how it should be.
[/b]
So true. Broadband is meaningless with capping.
A limited broadband is only slightly betterthan dial up IMO.
QUOTE(netfreak @ Oct 2 2006, 11:06 AM) [snapback]63860[/snapback]
So, WiFi adss less then 1% in terms of latency.
OTOH, this what IEEE specs say on VOIP QOS:
But overall, since default ping times are > 200 ms, you will never get POTS quality transmission irrespective of codec. If yo uswitch to a TCP based proxy instead of IP based, you can get POTS "sound" quality but that will have frustating delay due to TCP overhead.
[/b]
Umm...I am not aiming at sound quality. I hope to get decent sound quality without the "cracking" or "breaks" in the voice. What my current connection is offering is good sound quality but my voice breaks a lot. May be thats because of my upload speeds / latency? I am able to hear the calling party very clearly.
So, WiFi adss less then 1% in terms of latency.
OTOH, this what IEEE specs say on VOIP QOS:
But overall, since default ping times are > 200 ms, you will never get POTS quality transmission irrespective of codec. If yo uswitch to a TCP based proxy instead of IP based, you can get POTS "sound" quality but that will have frustating delay due to TCP overhead.
[/b]
Umm...I am not aiming at sound quality. I hope to get decent sound quality without the "cracking" or "breaks" in the voice. What my current connection is offering is good sound quality but my voice breaks a lot. May be thats because of my upload speeds / latency? I am able to hear the calling party very clearly.
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