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NEW QUESTION: 1
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. MGCP
B. SIP
C. RTP
D. H.323
Answer: C
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.
NEW QUESTION: 2
A. Option B
B. Option D
C. Option A
D. Option C
Answer: C
NEW QUESTION: 3
What are two possible options for representing a Web application within Internet Information Services (IIS)? (Each correct answer presents a complete solution. Choose two. )
A. Application server
B. Web site
C. Web directory
D. Virtual directory
E. Application directory
Answer: B,D
Explanation:
* Create a Web Application
An application is a grouping of content at the root level of a Web site or a grouping of content in a separate folder under the Web site's root directory. When you add an application in IIS 7, you designate a directory as the application root, or starting point, for the application and then specify properties specific to that particular application, such as the application pool that the application will run in.
* You can make an Existing Virtual Directory a Web Application.
NEW QUESTION: 4
The network administrator has been asked to give reasons for moving from IPv4 to IPv6.
What are two valid reasons for adopting IPv6 over IPv4? (Choose two.)
A. autoconfiguration
B. NAT
C. no broadcast
D. change of destination address in the IPv6 header
E. Telnet access does not require a password
F. change of source address in the IPv6 header
Answer: A,C
Explanation:
Explanation/Reference:
Explanation:
Six Benefits Of IPv6
http://www.networkcomputing.com/ipv6/six-benefits-of-ipv6/230500009
With IPv6, everything from appliances to automobiles can be interconnected. But an increased number of IT addresses isn't the only advantage of IPv6 over IPv4. In honor of World IPv6 Day, here are six more good reasons to make sure your hardware, software, and services support IPv6.
More Efficient Routing IPv6 reduces the size of routing tables and makes routing more efficient and hierarchical. IPv6 allows ISPs to aggregate the prefixes of their customers' networks into a single prefix and announce this one prefix to the IPv6 Internet. In addition, in IPv6 networks, fragmentation is handled by the source device, rather than the router, using a protocol for discovery of the path's maximum transmission unit (MTU).
More Efficient Packet Processing
IPv6's simplified packet header makes packet processing more efficient. Compared with IPv4, IPv6 contains no IP-level checksum, so the checksum does not need to be recalculated at every router hop.
Getting rid of the IPlevel checksum was possible because most link-layer technologies already contain checksum and error-control capabilities. In addition, most transport layers, which handle end-to-end connectivity, have a checksum that enables error detection.
Directed Data Flows IPv6 supports multicast rather than broadcast. Multicast allows bandwidth-intensive packet flows (like multimedia streams) to be sent to multiple destinations simultaneously, saving network bandwidth.
Disinterested hosts no longer must process broadcast packets. In addition, the IPv6 header has a new field, named Flow Label, that can identify packets belonging to the same flow. Simplified Network Configuration Address auto-configuration (address assignment) is built in to IPv6. A router will send the prefix of the local link in its router advertisements. A host can generate its own IP address by appending its link-layer (MAC) address, converted into Extended Universal Identifier (EUI) 64-bit format, to the 64 bits of the local link prefix.
Support For New Services
By eliminating Network Address Translation (NAT), true end-to-end connectivity at the IP layer is restored, enabling new and valuable services. Peer-to-peer networks are easier to create and maintain, and services such as VoIP and Quality of Service (QoS) become more robust.
Security IPSec, which provides confidentiality, authentication and data integrity, is baked into in IPv6.
Because of their potential to carry malware, IPv4 ICMP packets are often blocked by corporate firewalls, but ICMPv6, the implementation of the Internet Control Message Protocol for IPv6, may be permitted because IPSec can be applied to the ICMPv6 packets.