A new type of time-division multiplexing over IP (TDMoIP) will bring carrier-class quality levels to IP telephony. TDMoIP is a network that plugs into the traditional TDM network as a network that can seamlessly connect with existing equipment (such as traditional subscriber telephone exchanges and switches), and provide the same multiple services as traditional telephones, as well as quality assurance like the public switched telephone network (PSTN).
1、VoIP
The basic principle of VoIP's IP telephony is: firstly, the voice of the PCM is placed in the memory buffer of the IP gateway, and the gateway plays the role of protocol conversion, which is responsible for converting the PCM voice data stream into a compressed IP packet data stream. The buffers in the gateway are sampled from the PCM using a standard compression algorithm, which is sampled to form a discrete binary stream of data with a header. In the process of forming binary data, any redundant data, such as pauses (mutes) and redundancies between voices, will be flagged and may be compressed.
IP packets queued in the IP gateway buffer are routed and sent to their destination. When a packet arrives at the destination gateway, it is decompressed (re-inserted into the silence period and redundant voice) and decoded, restoring the original voice signal. The IP gateway plays the role of control and gatekeeping, such as call control, call management, network management, voice intelligent switching, etc.
Each IP gateway has a gateway number, that is, a unique IP address, which can be understood as the telephone office number or area code.
After dialing in the existing telephony equipment, the IP phone receives the call request through the local IP gateway, finds the IP address of the destination gateway number from its phone directory database, and makes a call request to the destination.
The main protocols used for call control similar to signaling in VoIP are the original CCITT H.323, H.225, H.245, etc.
VoIP to compete with the TDM networks of traditional telephony requires efforts on QoS assurance and signaling issues. In the voice service, low latency and correct signaling timing are crucial so that even a few milliseconds of signal loss can have little impact. To solve this problem, tunneling and jitter caching techniques can be employed. In addition, echo cancellation and voice compression technologies related to voice quality are not critical for data networks, but are needed for VoIP.
The basic functions of signaling are off-hook, ringing, correctly delivering the signal to the destination, and billing. Also related to signaling is calling user identification, call forwarding, conference phones, intelligent networking, etc. Users are often unaware of the multitude and complexity of the performance of the telephone network, and it is often felt by the user when some of the performance that the user is accustomed to is lost. This requires VoIP to also deal with the interface and coordination between the IP network and the traditional telephone network.