Internet Bandwidth consumed in making VoIP calls

This one is bit technical. As many of us have started making voice calls on the IP in our smartphones, I thought many of you would be interested to know how much internet bandwidth is consumed if you make a one minute of Voive call on IP i.e. VoIP call. Caveat - Some of highly technical stuff required cross-authentication from Geeks  (Learnt this from Wiki)

Here you go.....

Number of Bytes Consumed in VOIP

To know exactly how many bytes a voice conversation consumes, one needs to know which codec her VoIP service is using. 

CODECS 

A codec is a compression engine that transforms your (analog) voice into digital data, removing the silent moments (which make up to half of all conversations), and do other things to render the data load as light as possible. A codec is an algorithm (sort of a computer program), most of the time installed as a software on a server or embedded within a piece of hardware (IP Phones), that is used to convert voice (in the case of VoIP) signals into digital data to be transmitted over the Internet or any network during a VoIP call.

The word codec comes from the composed words coder-decoder or compressor-decompressor. Codecs normally achieve the following three tasks :

• Encoding – decoding
• Compression – decompression
• Encryption - Decryption

Encoding - decoding

Codec digitizes your analog voice to digital. When you talk over normal landline phone, your voice is transported in an analog way. But with VoIP, your voice is converted into digital signals. This conversion is technically called encoding, and is achieved by a codec. At destination, digitized voice is to be decoded back to its original analog state so that the other person can hear and understand it.

Compression – decompression

Bandwidth is a scarce commodity. To make the digitized voice less bulky, it is compressed. By compression, the same data is stored but using lesser space (digital bits). Once it reaches its destination, it is decompressed back to it original state before being decoded. 

Encryption – decryption

Encryption is one of the best tools for achieving security. It is the process of changing data into such a state that it no one can understand. This way, even if the encrypted data is intercepted by unauthorized people, the data still remains confidential. Once the encrypted data reaches destination, it is decrypted back to its original form. Often, when data is compressed, it already is encrypted to a certain extent, since it is altered from its original state.

Common VoIP Codecs

Codec	Bandwidth/kbps	Comments
G.711	64	Delivers precise speech transmission. low processor requirements. Needs at least 128 kbps for two-way.
G.722	48/56/64	Adapts to varying compressions
G.723.1	5.3/6.3	High compression with high quality audio.  Lot of processor power.
G.726	16/24/32/40	An improved version of G.721 and G.723 (different from G.723.1)
G.729	8	Excellent bandwidth utilization. Error tolerant. License required.
GSM	13	High compression ratio. Free and available in many hardware and software platforms.
iLBC	15	Robust to packet loss. Free
Speex	2.15 / 44	Minimizes bandwidth usage by using variable bit rate.

The approximate values for data consumption of the most common codecs used for VoIP are as follows:

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)

Note  - The G.729 codec is one of the best performing voice codecs and most good VoIP services use it.

Calculation of Bandwidth required

Example - for one minute of talk with the G.729 codec.

Now G.729 uses 8kpbs of sampling. Normally one packet of 20 Bytes is sent. With a sampling rate of 8kpbs this means that in one sec, 50 packets are to be sent.

If we are using frame relay protocol over ethernet, the overheads to be used are as follows - 

Frame relay - 6 Bytes(incl header and trailer)

RTP - 40 bytes (breakup 20 bytes of IP, RTP-12 & UDP - 8) [Note if cRTP is used these 40 Bytes can be converted into 4 Bytes]

Ethernet - 18 bytes

Therefore, for each packet of 20 bytes in this case 64 Bytes of protocol and other header bytes are added. Therefore total 84 Bytes will be send. Overall 50 such packets are semt per sec, therefore bit rate is 50*84*8= 33.6 kbps (Mentioned as ~32 kpbs)

This translates into 1920 kilobits (60 x 32) in one minute, which in turn is 240 kilobytes (KB) per minute (1 byte is 8 bits). Now that’s only for the data going out. Inbound data (which also counts) takes the same load, so one can double the figure to 480 KB. This gives a round value to 0.5 MB per minute of conversation.

However, there are many parameters, that are rather technical in nature, affecting the values above. Among them are the compression technology, the size (payload) of the voice packets, the intervals at which they are sent, packet overhead, network protocol used and whether silence suppression is used. (Pl refer table below). If silence suppression/voice activity detection is used, the bandwidth consumption may drop 50%

Packet voice transmission requirements (Bits per second per voice channel)
Codec	Voice bit rate	Sample time	Voice payload	Packets per second	Ethernet	PPP or Frame Relay RTP cRTP
G.711	64 Kbps	20 msec	160 bytes	50	87.2 Kbps	82.4 Kbps	68.0 Kbps
G.711	64 Kbps	30 msec	240 bytes	33.3	79.4 Kbps	76.2 Kbps	66.6 Kbps
G.711	64 Kbps	40 msec	320 bytes	25	75.6 Kbps	73.2 Kbps	66.0 Kbps
G.729A	8 Kbps	20 msec	20 bytes	50	31.2 Kbps	26.4 Kbps	12.0 Kbps
G.729A	8 Kbps	30 msec	30 bytes	33.3	23.4 Kbps	20.2 Kbps	10.7 Kbps
G.729A	8 Kbps	40 msec	40 bytes	25	19.6 Kbps	17.2 Kbps	10.0 Kbps
Note: RTP assumes 40-octets RTP/UDP/IP overhead per packet Compressed RTP (cRTP) assumes 4-octets RTP/UDP/IP overhead per packet Ethernet overhead adds 18-octets per packet PPP/Frame Relay overhead adds 6-octets per packet

If we take the value of 50 kbps for any codec, it will give (after calculations and approximations) 0.75 MB per minute of conversation.

VoIP revenues in North America to increase by 1431% by 2009 !

according to an Infonetics Research report , Voice-over-IP service revenues are poised for phenomenal growth in North America . That report predicts a growth rate of 1431% for VoIP service revenues, as they increase from $1.3 billion in 2004 to $19.9 billion in 2009.
That increase is predicted despite the fact that most businesses recently surveyed by Infonetics expect to provide their own VoIP, using a service provider pipe and premises-based VoIP gear.
VoIP revenues in 2004 represented less than 1% of total wireline carrier revenue.

Many service providers are still trying to figure out how to structure their hosted voice offerings, as well as how to bill for, manage and secure VoIP services. Particularly in courting business VoIP customers, service providers have to develop a mix of offerings that takes into account both hosted services and managed IP-PBX offerings. Service providers must be able to mix and manage customer premises solutions with hosting solutions even for one customer, since a business often has small offices that need a hosted solution, but also need the same feature set as the premises solution at corporate headquarters.
Service providers also are trying to sort out what falls into the flat-rate “all you can eat” category of voice service and what advanced features will generate badly needed revenue or create competitive differentiation.
Single office/home office (SOHO) users will be a major growth category for VoIP, the report predicts, growing from 1.1 million in 2004 to 20.8 million in 2008. Business-hosted VoIP service revenue is getting off to a slower start than that of managed IP-PBX services, but is expected to overtake those revenues by 2006 due to significant growth in the small to mid-sized market, according to Infonetics.

Ofcom, the UK regulator comes out with rules for VoIP telephony

U.K. regulator Ofcom has handed down its first set of rules for VoIP telephony - what it is calling a "code of practice" . The rules apply to "softphone" VoIP calls from PCs, such as those between Skype and landline users (via the SkypeOut service), as well as to calls using broadband-connected VoIP adapters and standard phones. In addition to Skype, which is thought to be as wildly popular in the U.K. as it is in most of Europe, major VoIP players in the U.K. include everyone from pure VoIP player Vonage to incumbent BT and major challengers such as France Telecom's Orange and Tesco.

Ofcom said that "the new code of practice requires VoIP providers to make clear:
>>Whether or not the service includes access to emergency services;
>>The extent to which the service depends on the user's home power supply;
>>Whether directory assistance, directory listings, access to the operator or the itemisation of calls are available; and
>>Whether consumers will be able to keep their telephone number if they choose to switch providers at a later date."
In cases where emergency service calls are not available - which in the U.K. is believed to me the vast majority of VoIP service - providers are being required the get consumer's positive written acknowledgement that they have been so informed. Ofcom suggested that getting them to check off a box on their service order could have that effect. VoIP providers also have to put stickers on VoIP phones warning users they can't make emergency calls and, should they try, have a recorded announcement to that effect. For softphones the warnings have to appear on the PC screen. Similar rules cover the issue of loss of service in a power outage.

Ofcoms' regulations hardly come as a surprise to the VoIP community. VoIP service providers had been fighting for years against Ofcom establishing any rules, a battle they have now lost. Ofcom began considering VoIP rules as far back as September 2004, the regulator noted in its ruling. A little over a year ago, in Feb. 2006, the it put forth its proposed rules.

While imposing its first regulations on VoIP, Ofcom also very carefully indicated its general approval of the emergence of the technology.
"VoIP services offer consumers the prospect of cheaper calls - especially for calls from one VoIP service to another - and valuable new services such as call handling and unified messaging," Ofcom wrote in its ruling announcement. "Over the last twelve months a range of new VoIP services have been launched and demand continues to grow."

The regulator also cited industry estimates that, by the end of this year, there will be some 3 million VoIP users in the U.K. That estimate may be quite low, and may not include softphone users such as those on Skype and Microsoft Messenger. In January BT put out a statement bragging of its one millionth VoIP customer, for instance, saying it hit the target a half a year earlier than it had expected.

Microsoft expects VoIP software market to reach $40b?

Microsoft is eyeing a shift in the way businesses use telephony and sees a move to web-based phone systems, or Voice over Internet Protocol [VoIP], according to Jeff Raikes, president of Microsoft's business division. It expects 100 million business users to make VoIP calls on its Office Communications software within three years. That is more than double the size of the current VoIP business market. Reuters reports only 8% of U.S. businesses have fully incorporated VoIP technology despite 82% of them using VoIP somewhere in their network, according to the Yankee Group. Cost has been a deterrent, but Microsoft says it expects a halving of prices in the next three years. Currently Microsoft faces business VoIP competition from Avaya and Cisco. Two versions of Microsoft's Office Communications software will be available for beta testing later this month.

Sources: Press release CNET, Reuters

5 Gbps transmission rate tested in 4G by DoCoMo

NTT DoCoMo announced Friday (9th Feb 07) that it achieved a maximum packet transmission rate of approximately 5Gbps in the downlink using 100MHz frequency bandwidth to a mobile station. .The field experiment of fourth-generation (4G) radio access took place in Yokosuka, Kanagawa Prefecture on December 25, 2006.
DoCoMo has already tested a maximum speed of 2.5Gbps on December 14, 2005.
The 5 Gbps speed was achieved by increasing the number of MIMO transmitting and receiving antennas from six to 12 each, and by using proprietary received signal processing technology.As compared with the December 14, 2005 test, the frequency spectrum efficiency, or the ratio of data transmission rate to channel bandwidth, was also doubled from 25bps/Hz to 50bps/Hz (5Gbps/100MHz).
The caveate as of now is that the test was done for mobile station moving only at speed fo 10km/h