Internet telephony uses the Internet to send audio between two or more computer users in the real time, so the users can converse. Vocaltec introduced the first Internet telephony software product in early 1995. Running a multimedia PC, the Vocaltec Internet Phone (and the numerous similar products introduced since) lets users speak into their microphones and listen via their speakers.
Two and half years passed, Internet telephony technology has caught the world's attention. The technology has improved to a point where conversations are easily possible. And it continues to get better. Dozens of companies have introduced products to commercialize the technology, and virtually every major telecommunications company has launched research to better understand this latest threat to their markets.
The second generation emerged after the development of technologies which overcame difficulties with PSTN interface protocols and the mapping of IP addresses to E.164 phone numbers. Using servers at the ISP's premises, these systems enable a user with a computer and an Internet connection to call any number on the PSTN.
The third generation phone gateways makes Internet telephony start to receive serious attention. These gateways provide a two-way interface between the PSTN and the Internet and allow voice conversations between users with standard phones, without the need of computers or Internet access.
Gateways are the key to bringing Internet telephony into the mainstream. By bridging the traditional circuit-switched telephony world with the Internet, gateways offer the advantages of Internet telephony to the most common, cheapest, most mobile, and easiest-to-use terminal in the world: the standard telephone. Gateways also overcome another significant Internet telephony problem, addressing. To address a remote user on a multimedia PC, you must know the user's Internet Protocol (IP) address. To address a remote user with a gateway product, you only need to know the user's phone number.
Right now, a common acceptable requirment for Internet Telephony
is:
the ability to establish a connection to the other end 95 percent of the time and a sound quality of at least 4-8 kbps, which is sub-toll quality.
The biggest difficulty that is facing VON (Voice On the Net) technology
is the interoperability between Internet telephony products and interworking
with legacy PSTN-based systems and services. Currently, no two products
are compatible. Users who want to make Internet phone call have to have
the same kind of software. Standard development and adoption are the key
to ensuring interoperability. At this time, the primary specification issues
that have to be resolved are related to the codec format, the transport
protocol, and directory services.
Voice performance is measured by delay. Calls on the public switched
telephone network usually exhibit 50- to 70-millisecond delay. That latency
increases substantially on the Internet, where it typically ranges from
500 milliseconds (an eternity when it comes to voice traffic). So right
now, some users still complain the quality when they use the Internet Telephony.
Latency affects the pace of the conversation. Humans can tolerate about 250msec of latency before it has a noticeable effect. Today's Internet telephony products exceed this latency, so most connections sound like traditional calls routed over a satellite circuit (which are usable, but require some getting used to). Even today, the products are well suited to many applications.
The Internet is an open network of many different ISPs' networks.
Consequently, there is no way to get network bandwidth, packet sequence
and latency guarantees. One of the main parameters affecting the quality
of service on the Internet is lost packets. Packet loss is a persistent
problem, particularly with the increasing popularity, and therefore increasing
load, of the Internet. Packet loss can occur for a number of reasons. Network
congestion due to bandwidth limitation or traffic overload is the main
reason. Inadequate network access links, especially local ISP connections
to the Internet backbone, are already causing chronic bandwidth congestion.
The heavy traffic loads are also straining the backbone infrastructure and leading to traffic collisions. The network overload also results in delays in packet transmission, with packets arriving too late at the receiver to be played back and are therefore, discarded. Congestion of routers and gateways also leads to packet discards. Error performance of underlying transmission paths is also an affecting factor with packet losses increasing dramatically on transcontinental links involving local ISPs and high error rate local networks. Finally, another reason for packet loss is the heavy loading of the servers leading to scheduling difficulties in multi-task operating systems. Current Internet telephony applications repair lost packets with silence, which leads to the speech clipping effects currently experienced by many Internet telephony users. Since comparatively large packets are used, even the loss of individual packets has a serious impact on the intelligibility of speech.
Regulation of Internet telephony is still largely a question mark.
Traditionally, telephone service has been heavily regulated. In most countries,
governments or government-sanctioned entities retain monopolies for provisioning
telephone service. Moreover, even without the Internet, telephony service
is deregulating in many countries around the world, although the deregulation
process is time consuming and heavily political.
Internet telephony has stirred fears from carriers throughout the globe, many of whom are reacting by seeking regulatory protection from the new technology. In the US, the America's Carriers Telecommunications Association (ACTA), a coalition of the smaller long distance carriers, filed a petition with the Federal Communications Commission (FCC).
ACTA argues that the major reason Internet telephone calls are cheaper
than traditional circuit-switched calls is the access charge exemption
ISPs enjoy. Hence, they argue, the Internet-based providers have an "unfair"
advantage in offering cut-rate long distance phone service. Fortunately
for the Internet telephone industry, and the Internet industry itself,
the FCC does not seem to agree.
The ITU H.323 recommendation which was ratified in May of last year,
defines the core technology for VON (Voice On the Net) applications. H.323
is based on the real-time protocol (RTP/RTCP) and is an extension of H.320
to include packet switched networks. H.323 is composed of a set of recommendations
including G.729 specifications for audio codecs, ratified by the ITU in
November 1995. The initial objective of the recommendation was to identify
a voice compression algorithm that could transport voice with quality equivalent
to 32 kbps ADPCM at only one-fourth of the bandwidth. The ratified standard
compresses signals to 8 kbps while delivering 4 kHz speech bandwidth with
toll quality. The adopted CS-ACELP algorithm meets stringent requirements
and objectives such as toll quality under clean channel conditions, robust
performance in the presence of random bit errors and detected erased frames.
In fact, the algorithm delivers an exceptionally high level of voice quality
with minimal delay and hopes are high that this state-of-the-art vocoder
will be adopted by major vendors.
H.323 also calls out T.120 for data conferencing. T.120 enables products from different vendors to interoperate without terminals assuming prior knowledge of the other systems. It specifies the network interfaces and wire formats, along with a data transmissions facility.
As for the transport protocol, RTP is finding acceptance as the standard means of transporting time related applications over the Internet. RTP has been introduced as a new protocol layer to provide support for applications with real-time properties including timing reconstruction, loss detection, security and content identification. RTP provides a time-stamp and control mechanisms for synchronising different streams with timing properties.
Since RTP does not address the issue of resource reservation or QoS control, it relies on the resource reservation protocol (RSVP) to provide these capabilities. Currently a draft standard protocol, the RSVP is part of various efforts to enhance the current Internet architecture with support for QoS flows to be able to handle real-time traffic more reliably. RSVP is a new signalling protocol to be implemented in Internet routers to provide for new classes of services by reserving paths for sessions on an end-to-end basis. This is achieved through three main functions, admission control, packet classification and packet scheduling. Each vendor has, however, its own strategy for performing these functions to provide the controlled load service at a first stage and the ultimate guaranteed service later. Although some vendors have already announced plans for providing telephony gateways with RSVP support, the reality is that after 13 consecutive revisions, RSVP specifications are still only a draft and testing of products and services for interoperability in public networks is still in the planning stages.
Serveral ways are used to improve the quality:
Right now, the average routers hop number of trans-Atlantic call
are 20 to 30. Since delay increase with each router hop, one solution is
to increase routing speed rather than putting in more routers. People prefer
the bigger routers is the solution to capacity. A gigarouter can handle
at least 10 times more traffic than a conventional router.
The per-packet cost of gigarouting is 3 to 4 times less than traditional
routing.
Although there are a lot of society issues, people always encourage
new technology. The trend of technology development can not be prevented
by anything. More than 100 companies are involed in the develpment of Internet
Telephony, including AT&T, MCI and Sprint. These long distance service
giants do not regard the Internet Telephony technology as their threats
but opportunities. Thinking about the supports from industry and keeping
promises that encourage competitions in 1996, FCC deny the petition from
ACTA and provides the offical support to Internet Telephony.
Right now, S.100 and other standards from the Enterprise Computer Telephony Forum are provided. They are important when building business communication platforms with integrated IP phone capability. Nearly 75% of U.S. companies already have or plan to incorporate intranets, according to Business Research Group (Newton, Mass.). So it can be forseen that in the near future, the Internet Telephony will work with Intranet together. It will come to most emploees' desk and everybody will familiar with them.
| Internet Telephony | Points | URL | |
| 1 | VocalTec Internet Phone | (1176) | http://www.vocaltec.com |
| 2 | TeleVox | ( 316) | http://www.voxware.com |
| 3 | NetMeeting | ( 314) | http://www.microsoft.com/netmeeting |
| 4 | Intel Internet Video Phone | ( 216) | http://www.intel.com |
| 5 | CU-SeeMe | ( 202) | http://www.wpine.com |
| 6 | CoolTalk | ( 173) | http://www.netscape.com |
| 7 | WebPhone | ( 169) | http://www.netspeak.com |
| 8 | FreeTel | ( 135) | http://www.freetel.com |
| 9 | VDOPhone | ( 111) | http://www.vdo.net |
| 10 | Net2Phone | ( 107) | http://www.net2phone.com |
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Last Modified: August 12, 1997