What is VoIP (Voice over Internet Protocol)?

VOIP refers to the use of IP protocol to transmit voice in the form of data packets on IP networks. Using the VOIP protocol, voice communication can be realized regardless of the Internet, intranet or LAN. In a network using VOIP, voice signals are digitized, compressed, and converted into IP packets, which are then transmitted over the IP network. The VOIP signaling protocol is used to set up and cancel calls, and transmit information needed to locate users and negotiate capabilities. The main characteristics of the telephone network are low cost; the synthesis of data, voice and video on the same network; new services on a centralized network and simplified management of end users.

VOIP protocol

Currently, there are some

In the past few years, the VOIP industry has been working on the following major issues:

Voice quality

Because IP is used to transmit data, it cannot provide real-time guarantee and can only provide the most effective service. In order for users to accept voice communications over IP, the packet delay needs to be less than a limit.

Interoperability

In a public network environment, products from different providers need to operate with each other to promote the wider application of VOIP.

safety

Use encryption (such as SSL) and tunnel (L2TP) technology to protect VOIP signaling and control traffic.

Integration of the Public Switched Telephone Network (PSTN)

-Although Internet telephony technology has been introduced, it needs to work with PSTN in the foreseeable future. Gateway technology is used to connect the two networks.

Scalability

VOIP needs to be flexible enough to grow with the growing private and public user markets. To solve the above problems, many network management and user management technologies are gradually being formed.

VOIP protocol

VOIP

Voice transmission protocols (also known as IP telephony, Internet telephony, and digital telephony) Paths for transmitting voice conversations through computers or other IP-based networks.

SIP

Session Initialization Protocol-an interactive user protocol developed by the IETF's MMUSIC working group and applied to standard initialization, modification, and termination of multimedia components such as video, voice, real-time information transmission, online gaming, and virtual reality.

PSTN

Public Switched Telephone Network-The World's Assembly of Public Lines-Switched Telephone Networks, just as the Internet is the meeting point for the transmission of public IP telephone data packets in the world.

ISDN

-Integrated Services Digital NetworkA type of network telephone conversion line system that transmits digital (as opposed to analog) voice and data through ordinary telephone copper wires to obtain better call quality and smoother transmission speeds than analog systems.

PBX

Internal corporate telephone exchange system (also known as corporate communication exchange system))))) a private enterprise-owned telephone transmission system, as opposed to a public carrier or telephone company-owned telephone transmission system

IVR

In the field of telephone technology, interactive voice responserefers to the computerized system allowing individuals, such as telephone callers, to follow the options in the voice menu, in other words, the individual interacts with the computer system.

DID

Direct dial-in (also known as European direct dial-in) is a client component provided by the telephone company, which is used for the communication exchange system between enterprises. The telephone company (telecommunications) collects all the enterprise communication switches (PBXs) connected to customers Number.

RFC

Request for Instructions (RFCs in the plural) is one of a series of widely-trained digital information files and standards for networks in commercial and free software in the Internet and Unix communities.

Traditional IP networks are mainly used to transmit data services. They use best-effort, connectionless technology, so there is no quality of service guarantee, packet loss, out-of-sequence arrival, and delay jitter. Data services do not have high requirements for this, but voice is a real-time service and has strict requirements on timing and delay. Therefore, special measures must be taken to ensure a certain quality of service. The key technologies of VOIP phone / VOIP network phone include: signaling technology, coding technology, real-time transmission technology, quality of service (QoS) guarantee technology, and network transmission technology.

Signaling technology

Signaling technology guarantees the smooth implementation of telephone calls and voice quality. Currently widely accepted VoIP control signaling systems include ITUT

H.323 series and IETF Session Initiation Protocol SIP.

The ITU H.323 series of recommendations define protocols and protocols for multimedia communications over the Internet or other packet networks without quality of service guarantees.

Its procedures. The H.323 standard is the technical foundation guarantee for multimedia on LAN, WAN, INTRANET and Internet.

H.323 is an ITU-T protocol suite for multimedia communications, including H.320 for ISDN and H.321 for B-ISDN

And H.324 recommendations for PSTN terminals. Its coding mechanism, protocol range and basic operations are similar to ISDN's Q.931 signaling

A simplified version of the protocol and uses a more traditional circuit-switched approach. Related protocols include H.245 for control, used for

Connected H.225.0, H.332 for large conferences, H.450.1, H.450.2, and H.450.3 for supplementary services.

Full H.235, H.246 interoperable with circuit switched services, etc. H.323 provides between devices, high-level applications, and providers

Interoperability. It does not depend on the network structure and is independent of the operating system and hardware platform. It supports multipoint functions, multicast and bandwidth management.

Management. H.323 has considerable flexibility to support conferences between nodes with different functions and conferences between different networks. H.323

The information flow in the proposed multimedia conference system includes audio, video, data and control information. Information flow adopts H.225.0 recommendation

Way to package and deliver.

The H.323 call establishment process involves three types of signaling: RAS signaling (R = Registration: Registration, A = Admission: Admission

And S = Status: Status), H.225.0 call signaling and H.245 control signaling. RAS signaling is used to complete the terminal and gatekeeper

Registration, authorization, bandwidth change, status, and release cancellation, etc .; H.225.0 call signaling is used to establish

Connection between two terminals, this signaling uses Q.931 messages to control the establishment and removal of calls. When there is no gatekeeper in the system, the call

The signaling channel is opened between the two terminals involved in the call; when the system includes a gatekeeper, the gatekeeper decides between the terminal and the gatekeeper

Open call signaling channels between or between two terminals; H.245 control signaling is used to transmit terminal-to-terminal control messages,

Including master-slave discrimination, capability exchange, opening and closing logical channels, mode parameter requests, flow control messages, and general commands and instructions.

The H.245 control signaling channel is established between two terminals, or between a terminal and a gatekeeper.

Although H.323 provides all the sub-protocols required for narrowband multimedia communications, the control protocols of H.323 are very complex. this

In addition, H.323 does not support the Multicast protocol. It can only use a Multipoint Control Unit (MCU) to form a multipoint conference.

At the same time, it can only support a limited number of multi-point users. H.323 does not support call forwarding, and it takes a long time to establish a call. With H.323

In contrast, SIP is a simpler session initialization protocol. It does not provide all communication protocols like H.323, but only provides

For session or call establishment and control. SIP can be used in areas such as multimedia conferencing, distance learning, and Internet telephony.

SIP supports both unicast and multicast. Session participants and media types can join an existing one at any time.

s meeting. SIP can be used to call people or machines, such as a media storage device to record a conference, or a point

The broadcast television server plays a video signal to the conference.

SIP is an application layer protocol, and UDP or TCP can be used as its transmission protocol. Unlike H.323: SIP is a basic

The text-based protocol is described in SIP Uniform Resource Locators, which is easy to implement and

Debugging is more important for flexibility and scalability. Because SIP is used only for initialization calls, not for transmitting media data,

The additional transmission cost is not large. SIP URLL can even be embedded in web pages or other hypertext links, users only need to use

With one click, you can place a call. Compared with H.323, SIP also has the characteristics of faster call establishment and support for transmitting phone numbers.

Coding technology

Voice compression coding technology is an important part of VOIP phone / VOIP network phone technology. Currently, the main coding techniques

There are G.729, G.723 (G.723.1), etc. as defined by ITU-T. Among them, G.729 can change the sampled 64kbit / s

The quality of the distortion is compressed to 8kbit / s. Because the quality of service cannot be well guaranteed in packet-switched networks, voice

Coding has certain flexibility, that is, variable adaptability of coding rate and coding scale. G.729 was originally a 8kbit / s voice editor

Code standard, the current working range is extended to 6.4 ~ 11.8kbit / s, the voice quality also changes within this range, but

K4kbit / s, the voice quality is also good, so it is very suitable for use in VoIP systems. G723.1 uses 5.3 / 6.3K bit / s dual rate words

Voice coding, its voice quality is good, but processing delay is large, it is the lowest rate voice coding algorithm that has been standardized at present.

Real-time transmission technology

The real-time transmission technology mainly uses the real-time transmission protocol RTP. RTP provides end-to-end real-time data transmission including audio

Agreement to send. RTP consists of data and control. The latter is called RTCP. RTP provides time stamping and control of different data streams.

The step-by-step mechanism allows the receiving end to reassemble the data packets of the sending end, and provides the quality of service of the receiving end to the multicast group.

Feed.

Quality of service (QoS) assurance technology

The VOIP phone / VOIP network phone mainly uses the Resource Reservation Protocol (RSVP) and real-time transmission control for service quality monitoring.

Protocol RTCP to avoid network congestion and ensure call quality.

Network Transmission Technology

The network transmission technology in VOIP phone / VOIP network phone is mainly TCP and UDP, in addition, it also includes gateway interconnection technology, routing

Selection technologies, network management technologies, and security authentication and billing technologies. Because the real-time transport protocol RTP provides

It is a proprietary, end-to-end data transmission service, so RTP can be used to transmit voice data in VOIP phones / VOIP Internet phones. in

The RTP header contains the identifier, sequence number, time stamp, and transmission monitoring of the loaded data. Generally, the RTP protocol data unit is

It is carried by UDP packets, and in order to minimize the delay, the voice payload is usually very short. IP, UDP, and RTP headers

Small length calculation. VoIP voice packets have a large overhead. The VOIP phone / VOIP network phone format using the RTP protocol is used in this way.

In the formula, multiple voices are inserted into the voice data segment, which improves the transmission efficiency. In addition, silence detection technology and echo cancellation technology also

It is a key technology in VOIP phone / VOIP network phone. Silent detection technology effectively eliminates silent signals, making speech

The occupied bandwidth of the signal is further reduced to about 3.5kbit / s; the echo cancellation technology mainly uses digital filter technology to eliminate the

Call quality has a great impact on echo interference, ensuring call quality.

VOIP phone

VoIP mobile phone is also called VoIP dual-mode mobile phone or IP mobile phone for short, it perfectly integrates GSM and WiFi, dual-mode standby at the same time, sharing user data. As an ordinary mobile phone, with it, users can use traditional G network services when there is no WiFi environment, including making calls, sending MMS, GPRS, etc .; wherever there is a wireless network, people can use it to enjoy free high-speed Surfing the web, IM chatting, sending and receiving e-mail, and other trendy mobile technologies. More importantly, people can use voip dual-mode mobile phones to call ordinary phones and mobile phones at a very low price through the network. It is free of charge and does not require the support of the operator. At the same time, three-way calling can be achieved through a mobile terminal. As a PDA mobile phone, the voip dual-mode mobile phone has many intelligent and thoughtful designs in use, including handwriting input, personal information management, reading and editing of common document formats, and shooting high-pixel digital pictures.

GSM / VoIP dual-mode smartphone (model 3000)

The Peritech dual-mode mobile phone is the world's first Window Mobile 6.0 dual-mode terminal product based on the SIP protocol. It has the following unique features:

TI OMAP 850 chipset

5 inch, 240x320 pixel TFT LCD

Memory (RAM: 64MB, ROM: 128MB)

2 million pixel CMOS camera

QWERTY full keyboard

Bluetooth 1.2 / IR 1.2

Windows Mobile 6.0 PPC

Tri-band: 900/1800 / 1900MHz

Support EDGE

Mini-SD memory card

GSM and VoWLAN stand by simultaneously

Can answer GSM and VoWLAN calls simultaneously