Which of the following protocols is used extensively in communication and entertainment systems that involve streaming media, such as telephony, video teleconference applications and web- based push to talk features?
A.
SIP
B.
MGCP
C.
H.323
D.
RTP
Explanation:
Real-time Transport Protocol (RTP), developed by the Audio-Video Transport Working Group of the IETF and first published in 1996, defines a standardized packet format for delivering audio and video over the Internet. RTP is used extensively in communication and entertainment systems that involve streaming media, such as telephony, video teleconference applications and web-based push to talk features. For these, it carries media streams controlled by H.323, MGCP, Megaco, SCCP, or Session Initiation Protocol (SIP) signaling protocols, making it one of the technical foundations of the Voice over IP industry. RTP is usually used in conjunction with the RTP Control Protocol (RTCP). When both protocols are used in conjunction, RTP is usually originated and received on even port numbers, whereas RTCP uses the next higher odd port number. RTP and RTCP typically use unprivileged UDP ports (1024 to 65535).Answer option C is incorrect. H.323 is a group of protocols defined by the International Telecommunication Union for multimedia conferences over Local Area Networks. The H.323 collection of protocols collectively may use up to two TCP connections and four to six UDP connections. H.323 inspection is used for H.323 compliant applications such as Cisco CallManager and VocalTec Gatekeeper. H.323 application inspecti
Answer option A is incorrect. Session Initiation Protocol (SIP), designed by Henning Schulzrinne and Mark Handley in 1996, is a signaling protocol, widely used for setting up and tearing down multimedia communication sessions such as voice and video calls over the Internet (VoIP). Other feasible application examples include video conferencing, streaming multimedia distribution, instant messaging, presence information and online games. The protocol can be used for creating, modifying, and terminating two-party (unicast) or multiparty (multicast) sessions consisting of one or several media streams. The modification can involve changing addresses or ports, inviting more participants, adding or deleting media streams, etc. The SIP protocol is a TCP/IP-based Application Layer protocol. Within the OSI model, it is sometimes placed in the session layer. SIP is designed to be independent of the underlying transport layer; it can run on TCP, UDP, or SCTP. It is a text-based protocol, sharing many elements of the Hypertext Transfer Protocol (HTTP) upon which it is based, allowing for easy inspection by administrators. SIP clients typically use TCP or UDP (typically on port 5060 and/or 5061) to connect to SIP servers and other SIP endpoints.
Answer option B is incorrect. MGCP stands for Media Gateway Control Protocol. The Media Gateway Control Protocol is architecture for controlling media gateways on Internet Protocol (IP) networks and the public switched telephone network (PSTN). It is a master/slave protocol used to control media gateways from external call control elements called media gateway controllers or call agents. A network element that provides conversion between the audio signals carried on telephone circuits and data packets carried over the Internet is called as media gateway. MGCP supports a large number of devices on an internal network with a limited set of external (global)
addresses using NAT and PAT.