a) SIP und H.323 b) SIP nutzt TCP bzw UDP (Transportschicht) und

a) SIP und H.323
b) SIP nutzt TCP bzw UDP (Transportschicht) und liegt in der Anwendungsschicht. Auf
der Netzschicht wird IPv4 oder IPv6 verwendet. H.323 nutzt diesselbe
Protokollstruktur.
c) A SIP message is either a request from a client to a server or a response from a server to a client. Both
the request and the response contain a start-line followed by one or more headers and a message body. For
example:
message = start-line
*message header
CRLF
[message-body]
The request line specifies the type of request being issued, while the response line indicates the success or
failure of a request. If a request is not executed,
d) ? (Client/Server??)
e) RTP (Real-time Transport Protocol)
f) An ADU descriptor consists of the following fields:
- "C": Continuation flag (1 bit)
- "T": Descriptor Type flag (1 bit)
– "ADU size" (6 or 14 bits)
g)
SIP requests are the codes used by Session Initiation Protocol for communication.
To complement them there are SIP Responses, which generally indicate whether this
request succeeded or failed, and in the latter case, why it failed.
* INVITE—Indicates a client is being invited to participate in a call session.
* ACK—Confirms that the client has received a final response to an INVITE request.
* BYE—Terminates a call and can be sent by either the caller or the callee.
* CANCEL—Cancels any pending searches but does not terminate a call that has
already been accepted.
* OPTIONS—Queries the capabilities of servers.
* REGISTER—Registers the address listed in the To header field with a SIP server.
SIP responses are the codes used by Session Initiation Protocol for communication.
They complement the SIP Requests, which are used to initiate action such as a
phone conversation. Note that the Reason Phrases of the responses listed below are
only the recommended examples, and can be replaced with local equivalents without
affecting the protocol.
1xx—Informational Responses
2xx—Successful Responses
3xx—Redirection Responses
4xx—Client Failure Responses
5xx—Server Failure Responses
6xx—Global Failure Responses
a) ?
b) * RTP – Real-Time Transport Protocol - defines a standardized packet format for
delivering audio and video over the Internet. RTP can carry any data with real-time
characteristics, such as interactive audio and video. Call setup and tear-down for
VoIP applications is usually performed by either SIP or H.323 protocols. The fact that
RTP uses a dynamic port range makes it difficult for it to traverse firewalls. In order to
get around this problem, it is often necessary to set up a STUN server.
* RTCP – Real-Time Control Protocol - provides out-of-band control information for
an RTP flow. It partners RTP in the delivery and packaging of multimedia data, but
does not transport any data itself. It is used periodically to transmit control packets to
participants in a streaming multimedia session. The primary function of RTCP is to
provide feedback on the quality of service being provided by RTP.
* RTSP – Real-Time Streaming Protocol - is a protocol for use in streaming media
systems which allows a client to remotely control a streaming media server, issuing
VCR-like commands such as "play" and "pause", and allowing time-based access to
files on a server.
The sending of streaming data itself is not part of the RTSP protocol. Most RTSP
servers use the standards-based RTP as the transport protocol for the actual
audio/video data, acting somewhat as a metadata channel. The RTSP server from
RealNetworks also features Real's proprietary RDT as the transport protocol.
c)
- RTP – Real-Time Transport Protocol
- RTCP – Real-Time Control Protocol
- RTSP – Real-Time Streaming Protocol
d) encryption and authentication
e) RTCP has five types of messages: sender report, receiver report, source description message, bye
message, application-specific message.
a) TCP, UDP, SCTP, DCCP ?
b) ?
c) ?
d) to improve TCP/IP performance over slow serial links
e) ?
f) Header compression reduces the normal 40 byte TCP/IP packet headers down to 3-4
bytes for the average case. It does this by saving the state of TCP connections at
both ends of a link, and only sending the differences in the header fields that change.
This makes a very big difference for interactive performance on low speed links,
although it will not do anything about the processing delay inherent to most dialup
modems.
a) Multicast Address-Set Claim (MASC)?
MASC is used by a node (typically a router) to claim and allocate one or more
address prefixes to that node's domain.
b) The multicast extensions to DHCP (MDHCP) provide configuration
parameters to the multicast applications. MDHCP is built on a
client-server model, where designated DHCP server allocates
multicast addresses and delivers parameters associated with the
address to dynamically configured hosts.
c)
* Host-to-router protocol (IGMP – Internet Group Management Protocol)
* Multicast routing protocols (various)
d) ?
e)
* Sourced-Based Distribution Tree
* Core-Based Distribution Tree
f) You use multicast scoping to limit multicast traffic by configuring it to an administratively defined
topological region. Multicast scoping controls the propagation of multicast messages—both multicast group
joins upstream toward a source and data forwarding downstream. Scoping can relieve stress on scarce
resources, such as bandwidth, and improve privacy or scaling properties.
a)  Optische Übertragungsebene
 Elektronische Übertragungsebene
 Vermittlungsebene
 Signalisierungsebene
 Ebene der Netzintelligenz
 Ebene des Netzmanagements
b)
Circuit Switching
Physical Connection
- Switched physical connection
- Isochronous
- Same bit rate at each end
- Constant end-to-end delay
- Exclusive usage of physical connection
Packet Switching
Logical Connection
- Switched logical connection
- Asynchronous or synchronous (realtime)
- Different or same bit rate at each end
- Variable end-to-end delay
- Shared usage of physical connection
c) ?
d) Play-out Puffer: zum Ausgleich der Verzögerungsschwankungen
e)
Tunnel-bypass by transmission paths:
- SDH (Synchronous Digital Hierarchy)
- WDM (Wavelength Division Multiplexing)
Constant delay
1) isochronous:
synchronous:
asynchronous:
2) ?
3) ?
4) ?
5) ?
1) ?
2) ?
3) Region 3: Überlast. Die Puffer in den Netzknoten füllen sich und bei Überlauf müssen
Pakete verworfen werden. Es gibt immer mehr Paketwiederholungen, die dazu
führen, dass das Netz immer stärker belastet wird, aber der eigentliche
Nutzdurchsatz abnimmt.
4)
Leaky Bucket
Token Bucket
- Buffer size determines allowed burst
size of data streams at input
- Token buffer size determines burst size
of data streams at output
- Data stream flows through leaky bucket
- Generator rate determines allowed data
rate
- Data loss at buffer overflow
- Data stream flows along Token output
- Equidistant data stream at output
- Data loss when no token available
- Smooth data stream at output
1) A: RTCP; B: RTSP; C: RTP; D: SRTP
2) encryption and authentication??
3) SIP?
4) RTP?
5) SIP requests are the codes used by Session Initiation Protocol for communication.
To complement them there are SIP Responses, which generally indicate whether this
request succeeded or failed, and in the latter case, why it failed.
* INVITE—Indicates a client is being invited to participate in a call session.
* ACK—Confirms that the client has received a final response to an INVITE request.
* BYE—Terminates a call and can be sent by either the caller or the callee.
* CANCEL—Cancels any pending searches but does not terminate a call that has
already been accepted.
* OPTIONS—Queries the capabilities of servers.
* REGISTER—Registers the address listed in the To header field with a SIP server.
SIP responses are the codes used by Session Initiation Protocol for communication.
They complement the SIP Requests, which are used to initiate action such as a
phone conversation. Note that the Reason Phrases of the responses listed below are
only the recommended examples, and can be replaced with local equivalents without
affecting the protocol.
1xx—Informational Responses
2xx—Successful Responses
3xx—Redirection Responses
4xx—Client Failure Responses
5xx—Server Failure Responses
6xx—Global Failure Responses
6) Stream Control Transmission Protocol
7)
Packet Header Size
12Bytes+Varible Chunk
Header
Reliability: Error recovery
by automatic repeat
request (ARQ)
YES
Virtual circuits: Sequence
numbering and reordering
Optional
Multiple Streams
YES
8)?
1) A = micro-mobility; B = macro-mobility; C = micro-mobility
2) ?
3) ?
4) ?
1) ?
2)
* Host-to-router protocol (IGMP – Internet Group Management Protocol)
* Multicast routing protocols (various)
3)
* Multicast routing protocol
4) ?
5) * Feedback Implosion
6)
* Clustering and Hierarchy
* Token
* Timers
a) Traditional: Isochronous, switched stream of 8-bit voice samples at a distance of
125 μs
Internet: Asynchronous, routed stream of packets with a group of 8-bit voice samples
b) Play-Out Buffer: sofern die Daten nicht echtzeitsensitiv sind. Also z.B. zeitversetztes
Video. Damit kann Delay Jitter durch Staus in Netzknoten ausgeglichen werden. Bei
echtzeitsensitiven Daten ist eine Ende-zu-Ende Verbindung nötig, die idealerweise
über WDM oder SDH Tunnel läuft, weil hier nur Hardware Jitter auftritt.
c) SIP und H.323
d) SIP nutzt TCP bzw UDP (Transportschicht) und liegt in der Anwendungsschicht. Auf
der Netzschicht wird IPv4 oder IPv6 verwendet. H.323 nutzt diesselbe
Protokollstruktur.
e) RTP
f) SIP requests are the codes used by Session Initiation Protocol for communication.
To complement them there are SIP Responses, which generally indicate whether this
request succeeded or failed, and in the latter case, why it failed.
* INVITE—Indicates a client is being invited to participate in a call session.
* ACK—Confirms that the client has received a final response to an INVITE request.
* BYE—Terminates a call and can be sent by either the caller or the callee.
* CANCEL—Cancels any pending searches but does not terminate a call that has
already been accepted.
* OPTIONS—Queries the capabilities of servers.
* REGISTER—Registers the address listed in the To header field with a SIP server.
SIP responses are the codes used by Session Initiation Protocol for communication.
They complement the SIP Requests, which are used to initiate action such as a
phone conversation. Note that the Reason Phrases of the responses listed below are
only the recommended examples, and can be replaced with local equivalents without
affecting the protocol.
1xx—Informational Responses
2xx—Successful Responses
3xx—Redirection Responses
4xx—Client Failure Responses
5xx—Server Failure Responses
6xx—Global Failure Responses
g)
* RTP – Real-Time Transport Protocol - defines a standardized packet format for
delivering audio and video over the Internet. RTP can carry any data with real-time
characteristics, such as interactive audio and video. Call setup and tear-down for
VoIP applications is usually performed by either SIP or H.323 protocols. The fact that
RTP uses a dynamic port range makes it difficult for it to traverse firewalls. In order to
get around this problem, it is often necessary to set up a STUN server.
* RTCP – Real-Time Control Protocol - provides out-of-band control information for
an RTP flow. It partners RTP in the delivery and packaging of multimedia data, but
does not transport any data itself. It is used periodically to transmit control packets to
participants in a streaming multimedia session. The primary function of RTCP is to
provide feedback on the quality of service being provided by RTP.
* RTSP – Real-Time Streaming Protocol - is a protocol for use in streaming media
systems which allows a client to remotely control a streaming media server, issuing
VCR-like commands such as "play" and "pause", and allowing time-based access to
files on a server.
The sending of streaming data itself is not part of the RTSP protocol. Most RTSP
servers use the standards-based RTP as the transport protocol for the actual
audio/video data, acting somewhat as a metadata channel. The RTSP server from
RealNetworks also features Real's proprietary RDT as the transport protocol.
h)
 RTP – Real-Time Transport Protocol
 RTCP – Real-Time Control Protocol
 RTSP – Real-Time Streaming Protocol
a)
* Stream Control Transmission Protocol
* Datagram Congestion Control Protocol
b)
DCCP
SCTP
Packet Header Size: Varies
12Bytes+Varible Chunk
Header
Reliability: Error recovery
by automatic repeat
request (ARQ):
No
Yes
Virtual circuits: Sequence
numbering and reordering:
Yes
Optional
Multiple Streams: No
Yes
c) to improve TCP/IP performance over slow serial links
d) Header compression reduces the normal 40 byte TCP/IP packet headers down to 3-4
bytes for the average case. It does this by saving the state of TCP connections at
both ends of a link, and only sending the differences in the header fields that change.
This makes a very big difference for interactive performance on low speed links,
although it will not do anything about the processing delay inherent to most dialup
modems.
e) Active Queue Management is a technique of preventing congestion in packetswitched
networks.
Algorithms:
• Adaptive Virtual Queue (AVQ);
• Random early detection (RED);
• Random Exponential Marking (REM);
• Blue and Stochastic Fair Blue (SFB);
• PI controller.
a)
* Host-to-router protocol (IGMP – Internet Group Management Protocol)
* Multicast routing protocols (various)
b) * Multicast routing protocol
c)
* Sourced-Based Distribution Tree
* Core-Based Distribution Tree
d) * Feedback Implosion
e)
* Clustering and Hierarchy
* Token
* Timers
f) DVB-S is the current digital television broadcast standard for satellite.
DVB-T is the current terrestrial digital television broadcast standard. It was developed
with the living room TV in mind, meaning that it best functions with a large rooftop
antenna, large TV screen, and a receiver (TV set) connected to a continuous power
supply. In contrast, DVB-H was developed with the handheld, mobile receiver in
mind.
In particular, DVB-H is optimized for use with battery-powered receivers with internal
antennas and small screens. DVB-H supports time-slicing technology, which
transmits the broadcast in bursts. The receiver can shut down between bursts, thus
saving power significantly.
a) Multicast Address-Set Claim (MASC)?
MASC is used by a node (typically a router) to claim and allocate one or more
address prefixes to that node's domain.
b) Multicast Address Dynamic Client Allocation Protocol (MADCAP)?
MADCAP is a protocol to enable multicasting in DHCP.
1)
Circuit Switching
Physical Connection
- Switched physical connection
- Isochronous
- Same bit rate at each end
- Constant end-to-end delay
- Exclusive usage of physical connection
Packet Switching
Logical Connection
- Switched logical connection
- Asynchronous or synchronous (realtime)
- Different or same bit rate at each end
- Variable end-to-end delay
- Shared usage of physical connection
2) optical transmission plane, electronic transmission plane, 64kbit/s, 2Mbit/s
3) wegen den Isonchronen Eigenschaften
4)
Layer 3
Packet Switching
Logical Channel
IP, X.25, GPRS, UMTS
Layer 2.5
Packet Switching
Logical Channel
MPLS
Layer 2
Packet Switching
Logical Channel
Ethernet, Frame Relay
Layer 1
Packet Swichting
Logical Channel
ATM
64 kbit/s, 2Mbit/s
Circuit Switching
Physical Channel
PSTN, ISDN, GSM, TDM
Electrical
Circuit Switching
Physical Channel
SDH, OTN
Optical
Circuit Switching
Physical Channel
WDM, OTDM
Optical
Packet Switching
Logical Channel
OBS, OPS
5) Max. 80ms erwünscht, 100-120ms ist tolerabel und über 200ms mühsame
Kommunikation.
6) It is a maximum value.
7) Play-out Puffer: zum Ausgleich der Verzögerungsschwankungen
8) Damit kann Delay Jitter durch Staus in Netzknoten ausgeglichen werden. Bei
echtzeitsensitiven Daten ist eine End zu End Verbindung nötig, die idealerweise über
WDM oder SDH Tunnel läuft, weil hier nur Hardware Jitter auftritt.
Siehe Punkt mit Delay Components -> Knotenverzögerung,...
9) Tunnel-bypass by switched end-to-end paths:
- MPLS (Multi-Protocol Label Switching)
- ATM (Asynchronous Transfer Mode)
Reduced delay jitter
10) Tunnel-bypass by transmission paths:
- SDH (Synchronous Digital Hierarchy)
- WDM (Wavelength Division Multiplexing)
Constant delay
11)
 Wired access (copper, coaxial cable, fiber)
 Access via PBX (Private Branch Exchange)
 Local area network (IEEE 802.3 Ethernet)
 Wireless local network (IEEE 802.11 WLAN)
 Mobile networks (GSM, UMTS)
Transmission limitation. Higher Delays.
12)
ADSL ~ 6Mbit/s,
FTTH GBit/s,
WLAN ~ 50Mbit/s,
UMTS ~ 250Kbit/s
13)
FTTH,
WLAN,
ADSL,
UMTS
1)






Optische Übertragungsebene
Elektronische Übertragungsebene
Vermittlungsebene
Signalisierungsebene
Ebene der Netzintelligenz
Ebene des Netzmanagements
2) Die Übertragungsendausrüstungen sind elektronisch und bilden eine eigene Ebene.
Die Übertragung zwischen Laser und Photodiode sowie die eventuelle optische
Vermittlung sind in der optischen Ebene angesiedelt.
3) Region 1: Linearer Anstieg bei steigendem Angebot.
Region 2: Beginn der Sättigung. Die Übertragungsleitungen arbeiten immer öfters mit voller Kapazität.
4) Region 3: Überlast. Die Puffer in den Netzknoten füllen sich und bei Überlauf müssen
Pakete verworfen werden. Es gibt immer mehr Paketwiederholungen, die dazu führen, dass das Netz immer
stärker belastet wird, aber der eigentliche Nutzdurchsatz abnimmt
5)
Leaky Bucket
Token Bucket
- Buffer size determines allowed burst
size of data streams at input
- Token buffer size determines burst size
of data streams at output
- Data stream flows through leaky bucket
- Generator rate determines allowed datarate
- Data loss at buffer overflow
- Data stream flows along Token output
- Equidistant data stream at output
- Data loss when no token available
- Smooth data stream at output
6) (1) mechanisms:
 IntServ (Admission Control, Packet Scheduler, Classifier, Reservationprotocol)
 DiffServ (PHB, Traffic Conditioners).
(2) IntServ (fine-grained, per-flow) & DiffServ (coarse-grained,per-aggregate)
(3) scalability: IntServ < DiffServ