VoIP Now: VoIP protocols

September 09, 2005

VoIP protocols - Part 3

SAP stands for Session Announcement Protocol. It is an announcement protocol used in the advertising of multicast media by the session directory clients. It is also used in communicating the session setup information to participants. The multicast announcement has the same scope as the session that it announces, this helps in keeping the local session announcement local and in maintaining the scalability of the protocol.

A SAP listener uses the Multicast-Scope Zone Announcement Protocol for listening to the multicast scopes on a SAP address and port. Instead of IPsec authentication headers, application level security is used in facilitate interoperability between mechanisms that are used for announcing the sessions. The session can be announced by a web page, a session initiation protocol, or by email.

The SAP protocol structure includes:

V: A version number field, which is three bits and is set to 1.
R: It stands for Reserved and is set to 0.
T: It is the message type and can have a value of 0 or 1, where 0 is the session announcement packet and 1 is the session deletion packet.
A: It is the Address Type and can have a value of 0 or 1. 0 is the originating source field and contains a 32-bit IPv4 address, 1 is the originating source and contains a 128-bit IPv6 address.
C: It is the compressed bit and the payload is compressed if C has a value of 1.
E: It is the encryption bit and can have a value of 0 or 1. 0 implies that the packet is not encrypted and that the timeout must be absent and 1 implies that the payload is encrypted and the timeout field has to be present at the packet header.
Timeout: It is a value that gives the NTP time for timing out a session and is included when the session payload has been encrypted and in the absence of the decryption key, listeners may not realize the timing fields in the payload.
Payload type: It specifies the MIME content type and it elaborates on the payload format.

SDP: It stands for Session Description Protocol and elaborates on session announcement and session invitation. A session directory tool present on the Internet Multicast Backbone (Mbone) helps in advertising the conference sessions and provides the conference address and other relevant information. The SDP messages, which are UDP packets, are relayed by multicasting an announcement packet to a popular multicast address using SAP. The messages carry a SAP header and a text payload and can be sent across the World Wide Web by using email. SDP uses different transport protocols such as SAP, SIP, RTSP, HTTP, etc. Also, SDP does not support session content negotiation and media encodings. The SDP messages consist of the session name, its duration, media details, and necessary information to access the media.

SIP: SIP stands for Session Initiation Protocol and it is an application-layer protocol that provides mechanisms for end user systems and proxy servers to establish, change, and end multimedia sessions including VoIP calls. It can also be used to initiate multicast conferences. Existing sessions can be modified by adding or removing media from it. Names can be mapped and services can be redirected with the help of SIP. This enables user mobility as a user can now have a single identifier independent of their location. SIP supports the following facets of multimedia communication:

User Location: It helps in determining the end system to be used in the communication process.
User Capabilities: Media parameters are determined by these.
User Availability: It checks for the readiness of the receiver to participate in a communication.
Session Management: It helps in modifying session parameters and ending sessions.
Session Setup: It sets up the session parameters at the caller's and the receiver's end.

SIP can be used as a component to develop a multimedia architecture like RTP that can be used to provide real-time data as well as feedback on QoS; the delivery of streaming media can be managed by RTSP; gateways to PSTN networks can be controlled by Megaco; and SDP can be used for providing information on the multimedia sessions. SIP can be used along with these protocols, however, its functioning is not impeded in the absence of these protocols. SIP is also used to provide security against DoS attacks, facilitate user to user and proxy to user authentication, and encryption.

In an Internet telephony session, SIP addresses are used to identify the caller and the receiver. A caller making an SIP call sends a request to the relevant server. The request may reach the receiver directly or it may lead to a number of SIP requests by the proxies. The SIP addresses can be present on web pages in the form of URLs, this helps in integrating them with applications like Click to talk.

The SIP messages can be sent using TCP and UDP, the messages are text based and use the UTF-8-encoded ISO 10646 character set. The messages are either requests or responses. The lines end with CRLF. An SIP request message consists of

Method: Methods include Invite, Ack, Options, Bye, etc and are carried out on the resource.
SIP version: The version of the SIP.
Request-URI: It is the SIP URL or the general Uniform Resource Identifier to which the request is addressed.

A response message header has the following format

Reason-phrase: It describes the status code.
SIP version: The version of the SIP
Status-code: It is an integer code that relates to the efforts to fulfill a request.

SGCP: It stands for Simple Gateway Control Protocol and it is an Internet protocol within a distributed system and is used to control telephony gateways, which are basically network elements that facilitate conversion between audio signals and data packets that are transferred over various networks. The SGCP works as a connection model and its two primary components are endpoints and connections. Call agents set up the connections that are grouped in calls. An endpoint consists of a domain name of a gateway and a local name inside the gateway.

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September 9, 2005 in Standards | Permalink