Overview of wireless VoIP technology

VoIP is a transmission technology that uses a router-Based IP group switching network to implement voice communication. The biggest advantage of VoIP technology is that IP Phones only require 8 kbit/s to use advanced voice encoding technology ~ The bandwidth of 12 kbit/s can be transmitted much less than the 64 kbit/s of the traditional circuit switching network. Wireless VoIP technology enables end-to-end voice transmission in wireless communication networks such as WLAN and UMTS over IP addresses. The main purpose of wireless VoIP technology development is to save bandwidth and improve spectrum utilization. The second is to provide unified interfaces and platforms for voice and data.

One of the Evolution Trends of 3G systems is network-wide IP. Full-IP network is mainly the IP address of the core network and client, which focuses on network structure IP, Protocol IP, and service IP, carrying, control, and business separation, in this way, the network will be more flexible. Due to the convenience of traditional voice services, voice services in 3G systems are still the most important and most important services. To achieve the IP address of the user end is not only to implement the integration of grouped data services on the user end, but also to implement the Packet Exchange bearer of voice over wireless access.

Since 1995, the VoIP technology has evolved from the initial PCtoPC simple voice communication to PCtoPhone, PhonetoPC, and PhonetoPhone commercial technologies. At the same time, the continuous expansion of data network capacity also helps VoIP technology in improving the quality of voice communication. Currently, the protocol layers for implementing real-time voice services over an IP network are roughly the same. They all use IETF's real-time transmission and control protocols. Different signaling protocols can be applied to call establishment and control, represented by ITU Standard Protocol H.323 and IETF Session Initiation Protocol (SIP.

The User Datagram Protocol (UDP) is selected at Layer 4th of the VoIP protocol stack to reduce the latency from one end to the other. However, UDP does not provide protection for data loss. The real-time transmission and control protocols over UDP provide information on the application layer of the voice encoding data packet sender. The information provided by the RTP data header includes the voice encoding scheme, serial number, voice sampling information, and the recognition information of the conversation audio data packets. RTCP is a signaling protocol that includes information about user interaction between sessions and provides feedback on session quality, such as lost RTP data packets, delay, and jitter at input intervals.

H.323 is a standard protocol cluster of ITU, this includes technical requirements for real-time interaction, such as video conferencing, data sharing, IP phones, and other multimedia communications, in a group network without service quality assurance. H.323 provides a description of the Multimedia Communication System and components of the data packet network, a description of the call method, and a call signaling program. Although H.323 is relatively complex and cumbersome in call control, it is still the most widely used VoIP Signaling System.

Unlike H.323, SIP provides all multimedia communication protocols. It only provides protocols related to call establishment and control functions. Compared with H.323, SIP is more flexible and easy to implement. It is expected to become an effective substitute for H.323, and SIP is the only call control protocol in 3GPPrelease5.

Because neither UDP nor RTP/RTCP can provide any VoIP service quality assurance, one way to provide high-quality services is to ensure sufficient bandwidth for packet transmission. Resource Reservation Protocol (RSVP) is used to send QoS requests to all nodes along the data flow, so that resources are reserved for any node participating in the session to establish and maintain the required data forwarding route. RSVP is a circuit switching solution that provides powerful QoS Assurance for VoIP.
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