Voice protocols – VoIP protocols

VoIP protocols functionalityThis article will list all well known and famous “voice protocols”. And, also we will shed some light on various things that those protocols are doing that make up a voice over IP network functional.
 

VoIP Protocols

VoIP networks are very popular these days. In order to support communication between traditional PBXs, Cisco IP phones, analog PSTN, and the analog telephones, all over IP network, quite a number of protocols are needed. Few protocols are indicating protocols (for instance, MGCP, H.323, SIP, H.248, and SCCP) used to position, sustain, and bring down a call. Other protocols are marked at the real voice packets (for example, SRTP, RTCP, and RTP) relatively indicating information. Few of the most common VoIP protocols are shown and described here.

  • SIP – This section is about “Session Initiation Protocol” (SIP), similar to H.323, it is regarded to be a “peer-to-peer” protocol! SIP is a famous, well known and popular protocol that can also be used and applied in a mixed-vendor surrounding, possibly due to its usage of existing and other residing protocols, such as SMTP and HTTP (Hyper text transfer protocol).
  • RTP – This section is related to “Real-time Transport Protocol” (RTP) this takes the payload of the voice. Fascinatingly, even though the RTP can be classified as a protocol that is of “Layer 4”, it is summarized in the internals of UDP (and this is also a protocol of Layer 4). Even though the port numbers of this UDP that are used; can have various differences regarding the vendor, particularly in Cisco surroundings, the “RTP” characteristically makes use of UDP ports within the range of 16,384 to 32,767.
  • RTCP – The control protocol of RTP (RTCP) supplies and provides with information regarding the flow of RTP (such as, information and data regarding the eminence of that call). According to context; in a Cisco surrounding or environment, RTCP characteristically makes use of odd UDP ports within the range 16,384-32,767.
  • SRTP – Secure RTP – protects the broadcast of the voice through RTP. Particularly and specifically, SRTP can add authentication, encryption, anti-replay mechanisms to voice traffic, and integrity.
  • H.323 – H.323 is an ITU customary. Instead of being just one protocol, it is a collection of protocols. Ahead of protocols, the H.323 customary also explains various devices, for example VoIP gatekeepers and VoIP gateways. H.323 is regarded as a peer-to-peer protocol; this is because few H.323 devices can create their very own call-routing choices, as contrasting to dependant on an outer catalog database.
  • MGCP – An initial development of Cisco, the “Media Gateway Control Protocol” (MGCP) is regarded as a client protocol or a server protocol as well. This server (for instance, has an analog port that is in a router with voice enabled feature) can also interact with a server (Cisco Unified Communications server is a common example in this criteria) through a cycle of signals and events. This particular server can tell the customer that in this event that compiles of a phone going off-hook starts the signal of dialing the tone to that phone.
  • H.248 – On the basis of MGCP, that is H.248 customary is also well known as Megaco in the technical nomenclature. Particularly, the new H.248 is actually a joint IETF and ITU customary. Even though H.248 also resembles the features of the MGCP, it is actually quite supple in terms of usage and its sustaining ability/to support for applications and gateways.
  • SCCP – This is about Skinny Client Control Protocol (SCCP), and it is frequently known as the skinny protocol, it is actually a proprietary signaling protocol of Cisco. SCCP is frequently and quite often used for the purpose of signaling (indicating) between and Cisco Unified Communications Manager servers Cisco IP Phones. But, various gateways of Cisco also help in sustaining SCCP. SCCP is regarded as a server/client protocol; examples include H.248 and also MGCP.

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