Voice-over-IP interface for standard household telephone

Details Voice-over-IP interface for standard household telephone Voice-over-IP interface for standard household telephone

2000-03-02

2004-11-30

Nguyen, Chau (Department: 2663)

Multiplex communications

Pathfinding or routing

Combined circuit switching and packet switching

C370S401000

Reexamination Certificate

active

06826174

ABSTRACT:

BACKGROUND OF THE INVENTION
1. The Field of the Invention
This invention is in the field of Voice Over Internet Protocol (VoIP) communications and, more particularly, to a system and method of interfacing a standard telephone to a VoIP compatible communication network.
2. The Prior State of the Art
Voice Over Internet Protocol (VoIP) is an emerging technology that allows the systems and wires that connect computer networks to act as an alternative to phone lines, delivering real-time voice to both standard telephones and PCs. VoIP allows an individual to utilize their computer connection to transmit voice encapsulated data packets over available local communication lines, such as the Internet, to another user on another computer, thereby creating a long distance phone call at a local connection price.
How VoIP Works
In a Voice-over-IP (VoIP) system, the analog voice signal is typically picked up by a microphone and sent to an audio processor within a PC. There, either a software or hardware CODEC performs analog-to-digital conversion and compression. Considerable research has been devoted to voice compression schemes that are well know to those skilled in the art. The nominal bandwidth required for telephone-type voice ranges from 2.9 Kbps (RT
24
by Voxware) to 13 Kbps (GSM cellular standard).
In placing the CODEC output into packets, there is a trade-off between bandwidth and latency. CODECs do not operate continuously. Instead, they sample the voice over a short period of time, known as a frame. These frames are like little bursts of data. One or more frames can be placed in a single IP datagram or packet, and then the packet payload is wrapped in the necessary packet headers and trailers. This packet overhead is at least 20 bytes for IP and 8 bytes for the User Datagram Protocol (UDP). Layer
2
protocols add even more overhead. Waiting longer to fill the IP datagram reduces overall overhead, which in turn reduces the true bandwidth needed to send the digitized voice. However, this waiting creates latency at the source, and too much total latency makes for a difficult conversation. Chart
1
shows the basic trade-off for initial latency versus true bandwidth.
The total network latency and jitter (changes in the latency) have a degrading effect upon voice quality. Therefore, real-time voice quality is difficult to maintain over a large wide-area packet network without priority handling. As previously mentioned, VoIP converts standard telephone voice signals into compressed data packets that can be sent locally over Ethernet or globally via an ISP's data networks rather than traditional phone lines. One of the main difficulties with VoIP connections is that the communication network supporting a VoIP platform must be able to recognize that VoIP data packets contain voice signals, and be “smart” enough to know that the communication network has to move the data packets quickly.
Presently, serious voice traffic does not use the public Internet but runs on private IP-based global networks that can deliver voice data with minimal congestion. As such, transmission of voice signals over private data networks offers businesses some great advantages. For ISPs, merging voice and data on one single network allows them to expand their services beyond simple information access and into the realm of voice, fax, and virtual private networking. For businesses, the benefit is big savings on long-distance service. The Internet right now is a free medium on many networks. If businesses can send voice over a computer network, businesses can conceivably make long-distance or international calls for the cost of a local call. VoIP further facilitates electronic commerce by allowing a customer service rep using one data line to answer telephone questions while simultaneously placing a customer's order online, perusing the company's web site, browsing an online information/product database, or sending an E-mail. Similarly, VoIP also creates new possibilities for remote workers, who for the cost of a local call can log in remotely, retrieve voice mail from their laptop PCs, and keep their E-mail and web applications running while conducting multiple voice and data calls over one phone line. Presently, this type of expanded VoIP functionality is exclusively limited to those with access to private IP based networks, such as business users and not the typical household user.
In fact, most household computer users are generally limited to the congested public Internet and cannot implement the VoIP standard effectively. If latency and jitter are too high, or the cost of reducing them is excessive, one alternative is to buffer the CODEC data at the receiver. A large buffer can be filled irregularly but emptied at a uniform rate. This permits good quality reproduction of voice. Such a buffering technique is known as audio streaming, and it is a very practical approach for recorded voice or audio. Unfortunately, excessive buffering of the audio signals leads to generally unacceptable one-sided telephone conversations, where one party dominates the transmissions. What is needed is a packetized telephone system that is able to compensate for latency and jitter, without introducing noticeable buffering.
Traditionally, the operating environment for a household user with a VoIP connection is either a laptop or desktop general-purpose computer. The recording and transmission or interpretation of the VoIP packets takes place in the sound system or modem DSP found on the laptop or desktop. As such, the desktop system has a minor advantage over the laptop, because the desktop sound system traditionally provides stereo surround speakers and an accurate microphone. Thus, the desktop system can more accurately capture an individual's voice for retransmission of these voice signals to the user on the other end of the connection. VoIP telephone software buffering and control structures help improve the connection, but even though the audio signal has been accurately sampled, the processor delays and transmission latency associated with the desktop VoIP connection over the public Internet tends to result in a barely audible VoIP call. What is needed is a household compatible packetized telephone system that is able to compensate for communication network delays and hardware limitations, without introducing noticeable degradation into the voice signal.
One of the main difficulties with using VoIP in a household system is that the protocol requires the user to follow numerous steps in order to establish a voice connection. In addition to the normal boot-up process associated with general-purpose computers for the operating system and the Internet telephone application, there are several details difficult for the household user to provide. For example, if a user were trying to contact another individual, they would need to know the individual's IP address and punch the address into their software application or web browser to contact the individual. Once the user contacts the individual through either E-mail or at the website, the user must notify them that the user wishes to initiate a VoIP connection. Then the individual being contacted would enable their VoIP to allow the user to begin streaming voice packets between the two devices. What is needed is a simple method of using VoIP with a household telephone, so that at the time the call is placed a user need only dial the access number on the telephone for the VoIP connection to be initiated and if possible connected.
In addition to the start-up delay and the awkward communication setup for most desktop systems, another problem with present VoIP systems is the immobility limitations imposed on the user by the VoIP desktop system. While the sound system is able to make an accurate recording, the user must sit at the desktop location or at least within range of the attached microphones and speakers to communicate. Unlike a telephone, the desktop system is very difficult to move to another room and it is generally considered very impractical to hav

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