SIP, H.323 and other protocols are used with VoIP. Internet data transmissions are composed of several layers. The network layer consists of the IP which establishes connection between two computers. The transport layer provides the rules that required for sending the data and the application layer determines how the data will be processed once it arrives at its destination.
Most data travelling over the Internet uses the Transmission Control Protocol (TCP) for the transport layer that gives guarantees data delivery and integrity. VoIP does not need that kind of delivery guarantee which TCP provides, so most VoIP reseller use transmissions use the faster User Datagram Protocol (UDP) as the transport layer.
Once VoIP data arrives at its destination, the application layer interprets it and presents it to the user. The most commonly used application layers for VoIP are SIP ( as VoIP Protocols- Part 1) and RTP.
Real-Time Transport Protocol (RTP) is the Internet protocol which transmits real-time data such as audio and video. RTP does not exclusively guarantee real-time delivery of data, but it provides mechanisms for sending and receiving applications to support streaming data.
As VoIP doesn’t use TCP (Transmission Control Protocol), RTP runs on top of the User Datagram protocol (UDP) instead. VoIP uses UDP as the transport layer. The UDP protocol provides direct method of sending and receiving data over an IP network and offers very few error recovery services. UDP has no mechanisms to notify the application of any loss in transmission whilst delivering packets of data; it also sends data unordered with no guarantees of the data being presented in the receiving application. All re-ordering of data into the correct format it was sent is hand VoIP Protocols – SIP and H.323:
There are a number of protocols that are used with VoIP. When looking for VoIP services and equipment, you may see references to SIP and H.323. These are the two most common protocols used for handling VoIP calls, but there are also many others protocols..
The Internet also uses numerous protocols, the most basic being IP (Internet Protocol). So what exactly are protocols? Protocols are simply a standard method (or set of rules) that must be followed in order for two or more devices to communicate with each other. Protocols can be anything from how a computer gets an IP address, to how a group of computers communicate across a network.
The Internet protocol (IP) allows devices connected to the internet to send and receive data. Data is sent back and forth in packets (or chunks) and contains source and destination data, timestamps, and if making a VoIP call, the actual digitised voice data.
Each device is distinctly identified by its IP address. The IP’s only concern is to send data across a network, but it offers little guarantee that the data will arrive intact. Other layers are used on top of IP in order to guarantee data integrity or speed of delivery.
When data integrity is important (for example when transmitting program files) a protocol like TCP (Transmission Control Protocol) is used on top of IP (also known as TCP/IP). However, the TCP is too slow for VoIP.
As VoIP depends on rapid delivery of data packets, it is not overly concerned if a few of the packets are dropped along the way. For this reason a number of different VoIP protocols have been developed. The two main protocols used for VoIP services are SIP and H.323.
SIP (Session Initiation Protocol) is a request-response protocol, dealing with requests from clients and responses from servers. It is becoming the standard for VoIP, and most VoIP service providers and soft phones use or at least offer this protocol. SIP was designed as a multimedia protocol that could take advantage of the architecture and messages found in popular Internet applications, such as voice, music and video. In addition to VoIP, SIP is used for videoconferencing and instant messaging.
When used for VoIP, SIP assigns each user a unique address. This address is independent of actual physical location, so the same SIP address can be used by one user anywhere in the world. SIP also defines standards for a number of different services including caller identification, conference calls, call forwarding, and user mobility.
As a Voice over IP protocol, SIP only defines how communication sessions are to be set up and torn down. It utilizes different protocols to define other aspects of VoIP and multimedia sessions, such as SDP for capabilities exchange, URLs for addressing, Domain Name Systems (DNSs) for service location, and Telephony Routing over IP (TRIP) for call routing.
H.323 is a protocol suite defined by ITU-T, used for voice transmission over Internet. In addition to voice applications, H.323 provides mechanisms for video communication and data collaboration.
H.323 was originally developed for multimedia streams over a Local Area Network, and was widely accepted in this arena. The standards of H.323 have been widely received and the specification continues to evolve. It is related to a suite of protocols which individually handle things like security, call signaling, and determining the capabilities of each party.
H.323 and SIP differ significantly in design, with H.323 being a binary protocol, and with SIP being an ASCII-based protocol. H.323 was developed before SIP, and seems to be losing ground to SIP as a standard VoIP Protocol. One reason for this is that SIP is much simpler than H.323. However, saying that, H.323 is still one of the major VoIP Protocols in use today, led by the RTP.