The VoIP is stand for Voice Over Internet Protocol. Simply it is transmission of voice traffic over IP based network.
Let's see some advantages of VoIP comparing with traditional analog telephone system.
Advantages;
- Integration of voice and data network
- Unified messaging of voice, email and fax messages
- Ability to communicate while not around the phone.
- Reduced cabling cost
- Reduced telephone company charges
- Preservation of analog technology
Generally the service provider provide the PSTN line. So when terminating this PSTN line need a analog voice interface. There are two type of analog voice interfaces;
- Foreign Exchange Station Interface (FXS)
- Foreign Exchange Office Interface (FXO)
FXS interface is connected directly to an analog endpoint such as an analog phone or fax machine withstandard RJ-11port. FXS ports are commonly found in residential homes that require very few analog lines. The interface provides voltage and signaling to analog devices. Following figure show a FXS port
Foreign Exchange Office Interface
Instead of plugging directly into an analog phone like the FXS port does, a FXO port connects to a PBX. The FXO interface assumes that all dial tones, ring indicators, and other call - progress signaling are provided locally by the equipment attached to it such as a key system or PBX.Contrast that with an FXS connection where the device that plugs into the port does not provide any form of signaling. Typically FXO ports from the PSTN and then connect the phones to the PBX. If an analog phone or fax machine needs to make an off - network call, the PBX switches the connection to one of the free analog lines.This is beneļ¬cial to businesses because not every phone requires an off - network analog line at any given time.
VoIP Endpoint Signaling Protocols
VoIP Endpoint Signaling Protocols
VoIP endpoint signaling protocols are responsible for locating endpoints, negotiation of various functions, and the setup and teardown of voice calls.
1. Session Initiation Protocol (SIP)
SIP is IETF standard signaling protocol and it is one of two supporting signaling protocols of Asterisk. The RFC for SIP states that the protocol was designed for the creation and management of multimedia sessions over the Internet. Its architecture is a peer-to-peer model in theory. SIP is a text based protocol and much similar to HTTP. SIP integrates well with other Internet applications such as email, instant messaging, and voice/video conferencing and collaboration.
The logical elements making up the SIP standard are the User Agent, Back-to-Back User Agent, Proxy Server, Redirect Server, and the Registrar. One big advantage of SIP is that these logical components can coexist with other applications on existing network components, making a costly infrastructure upgrade unnecessary.
2. H.323
H.323 was originally developed for videoconferencing over a packet based network, but was quickly adopted for Voice over IP. Its main function is to perform call control and management on an IP network. In H.323, for session set up, negotiation, and management, TCP is used for its reliability. But for time sensitive media such as voice and video, UDP is utilized as the transport mechanism because of its speed and low overhead.
3. Skinny Call Control Protocol (SCCP)
SCCP is a Cisco proprietary voice signaling protocol based on client-server architecture. So this is only work for any kind of Cisco end point phone.
4. Media Gateway Control Protocol (MGCP)
Media gateway control protocols were born out of the need for IP networks to inter work with traditional telephony systems and enable support of large-scale phone-to-phone deployments. Media gateway control protocols provide remote control of media streams as they transit between IP and traditional telephone networks.
Media gateway control protocols define how media streams are set up and establish media paths between IP and other network
5. Inter-Asterisk Exchange (IAX)
IAX was developed by Digium for the sole purpose of interconnecting between Asterisk servers. IAX is able to traverse signaling and media over a single User Datagram Protocol (UDP) port, while H.323 and SIP require a single UDP port for signaling and multiple dynamic UDP ports for media.
An added value is the ability to trunk multiple call sessions into a single dataflow. This feature preserves bandwidth and network resources when interconnecting Asterisk servers, utilizing the IAX protocol. Another advantage is security, as IAX is capable of performing authentication using plain text, MD5 hashing, and RSA key exchanges, making it more secure than Asterisk's SIP implementation.
Voice Codecs
Voice codecs are responsible for the encoding and decoding of voice signals. They
also can compress the digital signal so that more voice calls can be sent across a
limited amount of bandwidth.
1. G.711 ( u-law)
Voice codecs are responsible for the encoding and decoding of voice signals. They
also can compress the digital signal so that more voice calls can be sent across a
limited amount of bandwidth.
1. G.711 ( u-law)
G.711 is the fundamental codec of the PSTN. It also can be referred as PCM with respect to a telephone network. G.711 requires 64,000 bits to be transmitted per second. G.711 is an uncompressed codec. But it is hardly compressed G.711 imposes minimal (almost zero) load on the CPU
2. G.726
It is also known as Adaptive Differential Pulse-Code Modulation (ADPCM), and it can run at several bitrates. The most common rates are 16 Kbps, 24 Kbps, and 32 Kbps. But Asterisk currently supports only the ADPCM-32 rate.
G.726 offers quality nearly identical to G.711, but it uses only half the bandwidth.
3. G.729A
It uses very little bandwidth, but delivers impressive sound quality. G.729A uses Conjugate-Structure Algebraic-Code-Excited Linear Prediction (CS-ACELP).
To achieve its impressive compression ratio, this codec requires an equally impressive amount of effort from the CPU. G.729A is extremely popular codec, but without paying a licensing fee, the codec cannot be used.
4. GSM
It offers outstanding performance with respect to the demand it places on the CPU. The sound quality is generally considered to be of a lesser grade than that produced by G.729A. GSM operates at 13 Kbps.
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