Wednesday, May 18, 2005


VOIP (Voice over IP) Blog

Voice over IP (also called VoIP, IP Telephony, and Internet telephony) is technology enabling routing of voice conversations over the Internet or any other IP network. The voice data flows over a general-purpose packet-switched network, instead of the traditional dedicated, circuit-switched voice transmission lines.

This arrangement has several advantages over traditional telephony:

Freer innovation. Innovation progresses at market rates rather than the slow pace of the multilateral International Telecommunications Union (ITU) committee process, resulting in more new advanced features.
Lower per-call costs. Once a network connection is in place, a phone call may have no additional charge.
Lower infrastructure costs. VoIP reduces the traditional scheme—two separate wiring systems, one for voice and one for network—to a single connection.
Stability. A higher degree of reliability and resilience may be possible as network reliability improves.
"Future proof" hardware. Since functionality is software (protocol) based.
Protocols used to carry voice signals over the IP network are commonly referred to as Voice over IP or VoIP protocols.

Voice over IP traffic may be deployed on any IP network, including ones lacking an internet connection, for instance on a building-wide LAN without an internet connection.

Corporate and telco use of VoIP

Although few office environments and even fewer homes use a pure VoIP infrastructure, telecommunications providers routinely use IP telephony, often over a dedicated IP network, to connect switching stations, converting voice signals to IP packets and back. The result is a data-abstracted digital network which the provider can easily upgrade and use for multiple purposes. Corporate customer telephone support often use IP telephony exclusively to take advantage of the data abstraction. The benefit of using this technology is the need for only one class of circuit connection and better bandwidth use. IP telephony is commonly used to route traffic starting and ending at conventional PSTN (Public Switched Telephone Network) telephones. VoIP is widely employed by carriers, especially for international telephone calls. Electronic Numbering (Enum) uses standard phone numbers (E.164), but allows connections entirely over the Internet. If the other party uses Enum, the only expense is the Internet connection. Companies can acquire their own gateways, eliminating third-party costs — worthwhile in some situations.

Implementation challenges

Because IP does not provide any mechanism to ensure that data packets are delivered in sequential order, or provide any Quality of Service guarantees, VoIP implementations may face problems dealing with latency and data integrity.

A central challenge for VoIP implementers is restructuring streams of received IP packets, which can come in any order and have packets missing, to ensure that the ensuing audio stream maintains a proper time consistency. To help with this, the network provider can ensure that there is enough end-to-end bandwidth to guarantee low-latency, high quality voice. This is trivial in private networks, but very difficult with less than 256 kbit/s bandwith without a fragmentation mechanism.

Another main challenge is routing VoIP traffic to traverse firewalls and NAT. Intermediary devices called Session Border Controllers are often used to achieve this, though some proprietary systems such as Skype traverse firewall and NAT without a SBC.

Keeping packet latency acceptable on satellite circuits can also be a problem, simply due to transmission distances.

VoIP protocols

In the overwhelming majority of implementations, RTP is used to transmit VoIP traffic ("media"). The notable exception is IAX which carries both signaling and voice data over a UDP stream, which eases firewall and NAT traversal.

Signaling protocols:

Session Initiation Protocol (SIP)
an IETF newcomer gaining popularity
the ITU's widely deployed and continually updated VoIP protocol carrying billions of minutes of traffic each month
Skinny Client Control Protocol
proprietary protocol from Cisco
Megaco (a.k.a. H.248) and MGCP
both media gateway control protocols
proprietary protocol from Mitel
the Inter-Asterisk eXchange protocol used by the Asterisk open source PBX server and associated client software
Several different speech codecs can be used for stream audio compression. Commonly used codecs for VoIP traffic include G.711 and G.729, both ITU-T-specified codecs.

Mass-market telephony over broadband Internet access

A new development has been the introduction of mass-market VoIP services over broadband Internet access services, in which subscribers make and receive calls as they would over the PSTN. This requires an analog telephone adapter (ATA) to connect a telephone to the broadband Internet connection. Companies in the US, such as 1TouchTone, Broadvoice, Comcast, Verizon, Vonage, VoicePulse, Packet8 and SunRocket, use IP to offer unlimited calling to the US, and sometimes to Canada or to selected countries in Europe or Asia, for a flat monthly fee. One advantage of this is the ability to make and receive calls as one would at home, anywhere in the world, at no extra cost. As calls go via IP, this does not incur charges as call diversion does via the PSTN, and the called party does not have to pay for the call.

For example, somebody may call someone on a number with a US area code, but one could be in London, and if someone were to call another number with that area code, it would be treated as a local call, regardless of where that person is in the world. However, the broadband phone is likely to complement, rather than replace a PSTN line, as it still needs a power supply, while calling the US emergency services number 911, may not automatically be routed to the nearest local emergency dispatch center, or be of any use for subscribers outside the US.

Another challenge for these services is the proper handling of outgoing calls from fax machines, TiVo/ReplayTV boxes, satellite television receivers, alarm systems, conventional modems or FAXmodems, and other similar devices that depend on access to a voice-grade telephone line for some or all of their functionality. At present, these types of calls sometimes go through without a hitch, but in other cases they won't go through at all. And in some cases, this equipment can be made to work over a VoIP connection if the sending speed can be changed to a lower bits per second rate. If VoIP and cellular substitution becomes very popular, some ancillary equipment makers may be forced to redesign equipment, because it would no longer be possible to assume a conventional voice-grade telephone line would be available in almost all homes in North America and Western-Europe. The TestYourVoIP website offers a free service to test the quality of or diagnose an Internet connection by placing simulated VoIP calls from any Java-enabled Web browser, or from any phone or VoIP device capable of calling the PSTN network.

There is also a free service called Free World Dialup (FWD), that permits users to make free telephone calls to other FWD users, although has only limited connections to and from the public switched telephone network.


Blogger Crazy Dan said...

Hi Sanjay Das, I found your post VOIP Blog while doing some research on voip ip telephony. I like your blog, thank you for sharing the information and keep up the good work!. I really found some useful info!
I'll be back to see if you have any posts about voip ip telephony. Bye

7:54 am  
Blogger Tito Maury said...

Hi Sanjay Das, I found your blog while doing some research on explanation of voip. I like your blog, thank you for sharing the information and keep up the good work!.
I'll be back to see if you have any posts about explanation of voip

5:37 am  
Blogger bloger6us said...

Can anyone let me know of reliable free voip provider. What are the points that we have to be careful to choose a good free voip? There is so much information out there that it is very difficult to know with whom to go. I would appreciate any hekp on this. Thank you and good luck with your blog.

5:43 pm  
Blogger VOIP Provider said...

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3:37 am  

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