Tema 4: Aplicaciones Multimedia.
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Transcript Tema 4: Aplicaciones Multimedia.
Tema 3:
Protocolos de transporte multimedia.
Requisitos de la red
Gestión de los recursos: IntServ vs
DiffServ
RSVP
RTP/RTCP: Transporte de flujos
multimedia
RTSP: Control de sesión y localización de
medios
Protocolos para establecimiento y gestión
de sesiones multimedia
SIP
H.323
Asterisk
Thanks to :
RADCOM technologies
H. Shulzrinne
Paul. E. Jones (from packetizer.com)
Transmisión de Datos Multimedia –
http://www.grc.upv.es/docencia/tdm
– Master IC 2007/2008
Transmisión de Datos Multimedia - Master IC 2007/2008
Requisitos de red.
Se definen 3 parámetros críticos que la red debería
suministrar a las aplicaciones multimedia:
Productividad (Throughput)
Número de bits que la red es capaz de entregar por unidad
de tiempo (tráfico soportado).
CBR (streams de audio y vídeo sin comprimir)
VBR (ídem comprimido)
– Ráfagas (Peak Bit Rate y Mean Bit Rate)
Retardo de tránsito (Transit delay)
Mensaje listo
para envío
Envío del primer
bit del mensaje
Retardo
de acceso
2
Primer bit del
mensaje recibido
Ultimo bit del
Mensaje listo
mensaje recibido para recepción
Retardo
Retardo de
de tránsito
transmisión
Retardo extremo-a-extremo
Retardo
de acceso
Transmisión de Datos Multimedia - Master IC 2007/2008
Requisitos de red (II).
Varianza del retardo (Jitter)
Define la variabilidad del retardo de una red.
1
2
Emisor
3
t
1
D1
2
D2 = D1
3
Receptor
D3 > D1
t
Jitter físico (redes de conmutación de circuito)
– Suele ser muy pequeño (ns)
LAN jitter (Ethernet, FDDI).
– Jitter físico + tiempo de acceso al medio.
Redes WAN de conmutación de paquete (IP, X.25, FR)
– Jitter físico + tiempo de acceso + retardo de conmutación de paquete en conmutadores de la
3
red.
Transmisión de Datos Multimedia - Master IC 2007/2008
Internet y las aplicaciones multimedia
¿Qué podemos añadir a IP para soportar los
requerimientos de las aplicaciones multimedia?
Técnicas de ecualización de retardos (buffering)
Protocolos de transporte que se ajusten mejor a las
necesidades de las aplicaciones multimedia:
RTP (Real-Time Transport Protocol) RFC 1889.
RTSP (Real-Time Streaming Protocol) RFC 2326.
Técnicas de control de admisión y reserva de recursos
(QoS)
RSVP (Resource reSerVation Protocol) RFC 2205
Arquitecturas y protocolos específicos:
Protocolos SIP (RFC 2543), SDP (RFC 2327), SAP (RFC
2974), etc..
ITU H.323
4
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Internet Protocols
Slide thanks to Henning Schulzrinne
5
Tema 3:
Protocolos de transporte multimedia.
Requisitos de la red
Gestión de los recursos: IntServ vs
DiffServ
Computer Networking: A
Top Down Approach
Featuring the Internet,
RSVP
RTP/RTCP: Transporte de flujos
multimedia
RTSP: Control de sesión y localización
de medios
Protocolos para establecimiento y
gestión de sesiones multimedia
SIP
H.323
Transmisión de Datos Multimedia –
3rd edition.
Jim Kurose, Keith Ross
Addison-Wesley, July 2004.
Thanks to :
RADCOM technologies
H. Shulzrinne
Paul. E. Jones (from packetizer.com)
http://www.grc.upv.es/docencia/tdm
– Master IC 2007/2008
Transmisión de Datos Multimedia - Master IC 2007/2008
7
Scheduling And Policing Mechanisms
scheduling: choose next packet to send on link
FIFO (first in first out) scheduling: send in order of arrival to queue
discard policy: if packet arrives to full queue: who to discard?
Tail drop: drop arriving packet
priority: drop/remove on priority basis
random: drop/remove randomly
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Scheduling Policies: more
Priority scheduling: transmit highest priority queued packet
multiple classes, with different priorities
class may depend on marking or other header info, e.g. IP
source/dest, port numbers, etc..
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Scheduling Policies: still more
round robin scheduling:
multiple classes
cyclically scan class queues, serving one from each class (if
available)
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1
0
Scheduling Policies: still more
Weighted Fair Queuing:
generalized Round Robin
each class gets weighted amount of service in each cycle
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1
1
Policing Mechanisms
Goal: limit traffic to not exceed declared parameters
Three common-used criteria:
(Long term) Average Rate: how many pkts can be sent per unit time (in
the long run)
crucial question: what is the interval length: 100 packets per sec or 6000
packets per min have same average!
Peak Rate: e.g., 6000 pkts per min. (ppm) avg.; 1500 ppm peak rate
(Max.) Burst Size: max. number of pkts sent consecutively (with no
intervening idle)
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Policing Mechanisms
Token Bucket: limit input to specified Burst Size and Average Rate.
bucket can hold b tokens
tokens generated at rate r token/sec unless bucket full
over interval of length t: number of packets admitted less than or
equal to (r t + b).
1
2
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Policing Mechanisms (more)
token bucket, WFQ combine to provide guaranteed upper bound
on delay, i.e., QoS guarantee!
arriving
traffic
token rate, r
bucket size, b
WFQ
per-flow
rate, R
D = b/R
max
1
3
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IETF Integrated Services
architecture for providing QOS guarantees in IP networks for
individual application sessions
resource reservation: routers maintain state info (a la VC) of
allocated resources, QoS req’s
admit/deny new call setup requests:
Question: can newly arriving flow be admitted
with performance guarantees while not violated
QoS guarantees made to already admitted flows?
1
4
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Intserv: QoS guarantee scenario
Resource reservation
call setup, signaling (RSVP)
traffic, QoS declaration
per-element admission control
request/
reply
QoS-sensitive
scheduling (e.g.,
WFQ)
1
5
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1
6
Call Admission
Arriving session must :
declare its QOS requirement
R-spec: defines the QOS being requested
characterize traffic it will send into network
T-spec: defines traffic characteristics
signaling protocol: needed to carry R-spec and T-spec to routers
(where reservation is required)
RSVP
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Intserv QoS: Service models [RFC2211, RFC2212]
Guaranteed service:
worst case traffic arrival: leaky-bucketpoliced source
simple (mathematically provable) bound
on delay [Parekh 1992, Cruz 1988]
arriving
traffic
Controlled load service:
"a quality of service closely
approximating the QoS that same flow
would receive from an unloaded
network element."
token rate, r
bucket size, b
WFQ
per-flow
rate, R
D = b/R
max
1
7
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IETF Differentiated Services
Concerns with Intserv:
Scalability: signaling, maintaining per-flow router state difficult
with large number of flows
Flexible Service Models: Intserv has only two classes. Also want
“qualitative” service classes
“behaves like a wire”
relative service distinction: Platinum, Gold, Silver
Diffserv approach:
simple functions in network core, relatively complex functions at
edge routers (or hosts)
Don’t define define service classes, provide functional
components to build service classes
1
8
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Diffserv Architecture
Edge router:
per-flow traffic management
marks packets as in-profile
and out-profile
Core router:
per class traffic management
buffering and scheduling based
on marking at edge
preference given to in-profile
packets
Assured Forwarding
1
9
r marking
scheduling
b
..
.
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Edge-router Packet Marking
profile: pre-negotiated rate A, bucket size B
packet marking at edge based on per-flow profile
Rate A
B
User packets
Possible usage of marking:
class-based marking: packets of different classes marked differently
intra-class marking: conforming portion of flow marked differently than nonconforming one
2
0
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2
1
Classification and Conditioning
Packet is marked in the Type of Service (TOS) in IPv4, and Traffic
Class in IPv6
6 bits used for Differentiated Service Code Point (DSCP) and
determine PHB that the packet will receive
2 bits are currently unused
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2
2
Classification and Conditioning
may be desirable to limit traffic injection rate of some class:
user declares traffic profile (e.g., rate, burst size)
traffic metered, shaped if non-conforming
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2
3
Forwarding (PHB)
PHB result in a different observable (measurable) forwarding
performance behavior
PHB does not specify what mechanisms to use to ensure required
PHB performance behavior
Examples:
Class A gets x% of outgoing link bandwidth over time intervals of a
specified length
Class A packets leave first before packets from class B
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2
4
Forwarding (PHB)
PHBs being developed:
Expedited Forwarding: pkt departure rate of a class equals or
exceeds specified rate
logical link with a minimum guaranteed rate
Assured Forwarding: 4 classes of traffic
each guaranteed minimum amount of bandwidth
each with three drop preference partitions
Tema 3:
Protocolos de transporte multimedia.
Requisitos de la red
Gestión de los recursos: IntServ vs
DiffServ
Computer Networking: A
Top Down Approach
Featuring the Internet,
RSVP
RTP/RTCP: Transporte de flujos
multimedia
RTSP: Control de sesión y localización
de medios
Protocolos para establecimiento y
gestión de sesiones multimedia
SIP
H.323
Transmisión de Datos Multimedia –
3rd edition.
Jim Kurose, Keith Ross
Addison-Wesley, July 2004.
Thanks to :
RADCOM technologies
H. Shulzrinne
Paul. E. Jones (from packetizer.com)
http://www.grc.upv.es/docencia/tdm
– Master IC 2007/2008
Transmisión de Datos Multimedia - Master IC 2007/2008
Signaling in the Internet
connectionless
(stateless)
forwarding by IP
routers
+
best effort
service
=
no network
signaling protocols
in initial IP
design
New requirement: reserve resources along end-to-end path (end
system, routers) for QoS for multimedia applications
RSVP: Resource Reservation Protocol [RFC 2205]
“ … allow users to communicate requirements to network in robust and
efficient way.” i.e., signaling !
earlier Internet Signaling protocol: ST-II [RFC 1819]
2
6
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2
7
RSVP Design Goals
1.
2.
3.
4.
5.
6.
accommodate heterogeneous receivers (different bandwidth along
paths)
accommodate different applications with different resource
requirements
make multicast a first class service, with adaptation to multicast
group membership
leverage existing multicast/unicast routing, with adaptation to
changes in underlying unicast, multicast routes
control protocol overhead to grow (at worst) linear in # receivers
modular design for heterogeneous underlying technologies
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2
8
RSVP: does not…
specify how resources are to be reserved
rather: a mechanism for communicating needs
determine routes packets will take
that’s the job of routing protocols
signaling decoupled from routing
interact with forwarding of packets
separation of control (signaling) and data (forwarding) planes
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RSVP: overview of operation
senders, receiver join a multicast
group
done outside of RSVP
senders need not join group
sender-to-network signaling
path message: make sender
presence known to routers
path teardown: delete sender’s
path state from routers
receiver-to-network signaling
reservation message: reserve
resources from sender(s) to
receiver
reservation teardown: remove
receiver reservations
network-to-end-system signaling
path error
reservation error
2
9
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3
0
Call Admission
Session must first declare its QOS requirement and characterize the
traffic it will send through the network
R-spec: defines the QOS being requested
T-spec: defines the traffic characteristics
A signaling protocol is needed to carry the R-spec and T-spec to the
routers where reservation is required;
RSVP is a leading candidate for such signaling protocol
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RSVP request (T-Spec)
A token bucket specification
bucket size, b
token rate, r
the packet is transmitted onward only if the number of tokens in the
bucket is at least as large as the packet
peak rate, p
p>r
maximum packet size, M
minimum policed unit, m
All packets less than m bytes are considered to be m bytes
Reduces the overhead to process each packet
Bound the bandwidth overhead of link-level headers
3
1
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3
2
Call Admission
Call Admission: routers will admit calls based on their R-spec and Tspec and base on the current resource allocated at the routers to
other calls.
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3
3
Integrated Services: Classes
Guaranteed QOS: this class is provided with firm bounds on
queuing delay at a router; envisioned for hard real-time applications
that are highly sensitive to end-to-end delay expectation and
variance
Controlled Load: this class is provided a QOS closely
approximating that provided by an unloaded router; envisioned for
today’s IP network real-time applications which perform well in an
unloaded network
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3
4
R-spec
An indication of the QoS control service requested
Controlled-load service and Guaranteed service
For Controlled-load service
Simply a Tspec
For Guaranteed service
A Rate (R) term, the bandwidth required
R r, extra bandwidth will reduce queuing delays
A Slack (S) term
The difference between the desired delay and the delay that would be
achieved if rate R were used
With a zero slack term, each router along the path must reserve R
bandwidth
A nonzero slack term offers the individual routers greater flexibility in making
their local reservation
Number decreased by routers on the path.
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3
5
QoS Routing: Multiple constraints
A request specifies the desired QoS requirements
e.g., BW, Delay, Jitter, packet loss, path reliability etc
Two type of constraints:
Additive: e.g., delay
Maximum (or Minimum): e.g., Bandwidth
Task
Find a (min cost) path which satisfies the constraints
if no feasible path found, reject the connection
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Path msgs: RSVP sender-to-network signaling
path message contents:
address: unicast destination, or multicast group
flowspec: bandwidth requirements spec.
filter flag: if yes, record identities of upstream senders (to allow packets
filtering by source)
previous hop: upstream router/host ID
refresh time: time until this info times out
path message: communicates sender info, and reverse-path-tosender routing info
later upstream forwarding of receiver reservations
3
6
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RSVP: simple audio conference
H1, H2, H3, H4, H5 both senders and receivers
multicast group m1
no filtering: packets from any sender forwarded
audio rate: b
only one multicast routing tree possible
H3
H2
R1
R2
H1
H5
3
7
R3
H4
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RSVP: building up path state
H1, …, H5 all send path messages on m1:
(address=m1, Tspec=b, filter-spec=no-filter,refresh=100)
Suppose H1 sends first path message
m1:
m1:
in L1
out
L2 L6
in
L7
out L3 L4
L6
m1: in
out L5
L7
H3
H2
L3
L2
H1
L1
R1
L6
R2
L5
H5
3
8
L7
R3
L4
H4
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RSVP: building up path state
next, H5 sends path message, creating more state in routers
m1:
L6
L1
m1: in
out L1 L2 L6
in
L7
out L3 L4
L5 L6
m1: in
out L5 L6 L7
H3
H2
L3
L2
H1
L1
R1
L6
R2
L5
H5
3
9
L7
R3
L4
H4
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RSVP: building up path state
H2, H3, H5 send path msgs, completing path state tables
m1:
L1 L2 L6
m1: in
out L1 L2 L6
in L3 L4 L7
out L3 L4 L7
L5 L6 L7
m1: in
out L5 L6 L7
H3
H2
L3
L2
H1
L1
R1
L6
R2
L5
H5
4
0
L7
R3
L4
H4
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4
1
reservation msgs: receiver-to-network signaling
reservation message contents:
desired bandwidth:
filter type:
no filter: any packets address to multicast group can use reservation
fixed filter: only packets from specific set of senders can use reservation
dynamic filter: senders who’s packets can be forwarded across link will
change (by receiver choice) over time.
filter spec
reservations flow upstream from receiver-to-senders, reserving
resources, creating additional, receiver-related state at routers
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RSVP: receiver reservation example 1
H1 wants to receive audio from all other senders
H1 reservation msg flows uptree to sources
H1 only reserves enough bandwidth for 1 audio stream
reservation is of type “no filter” – any sender can use reserved
bandwidth
H3
H2
L3
L2
H1
L1
R1
L6
R2
L5
H5
4
2
L7
R3
L4
H4
Transmisión de Datos Multimedia - Master IC 2007/2008
RSVP: receiver reservation example 1
H1 reservation msgs flows uptree to sources
routers, hosts reserve bandwidth b needed on downstream links
towards H1
m1: in L1 L2
out L1(b) L2
L6
L6
m1:
L2
H1
b
b
L1
R1
b
L6
b
R2
L5
H5
4
3
L7
L7(b)
L7
L6
L6(b) L7
m1: in L5
out L5
H2
L4
L4
in L3
out L3
b
L7
b
R3
L3
b
L4
H3
H4
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RSVP: receiver reservation example 1 (more)
next, H2 makes no-filter reservation for bandwidth b
H2 forwards to R1, R1 forwards to H1 and R2 (?)
R2 takes no action, since b already reserved on L6
L6
m1: in L1 L2
out L1(b) L2(b) L6
m1:
b
L2
H1
b
b
b L1
R1
b
L6
b
R2
L5
H5
4
4
L7
L7(b)
L7
L6
L6(b) L7
m1: in L5
out L5
H2
L4
L4
in L3
out L3
b
L7
b
R3
L3
b
L4
H3
H4
Transmisión de Datos Multimedia - Master IC 2007/2008
RSVP: receiver reservation: issues
What if multiple senders (e.g., H3, H4, H5) over link (e.g., L6)?
arbitrary interleaving of packets
L6 flow policed by leaky bucket: if H3+H4+H5 sending rate exceeds b,
packet loss will occur
L6
m1: in L1 L2
out L1(b) L2(b) L6
m1:
b
L2
H1
b
b
b L1
R1
b
L6
b
R2
L5
H5
4
5
L7
L7(b)
L7
L6
L6(b) L7
m1: in L5
out L5
H2
L4
L4
in L3
out L3
b
L7
b
R3
L3
b
L4
H3
H4
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RSVP: example 2
H1, H4 are only senders
send path messages as before, indicating filtered reservation
Routers store upstream senders for each upstream link
H2 will want to receive from H4 (only)
H3
H2
L3
L2
H1
4
6
L1
R1
L6
R2
L7
R3
L4
H4
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RSVP: example 2
H1, H4 are only senders
send path messages as before, indicating filtered reservation
in
L1, L6
L2(H1-via-H1
out L6(H1-via-H1
L1(H4-via-R2
in
; H4-via-R2
)
)
)
L4, L7
L3(H4-via-H4
out L4(H1-via-R2
L7(H4-via-H4
)
H3
H2
R2
L2
H1
L1
R1
L7
L6
in
L3
R3
L6, L7
L6(H4-via-R3
out L7(H1-via-R1
4
7
; H1-via-R3
)
)
)
)
L4
H4
Transmisión de Datos Multimedia - Master IC 2007/2008
RSVP: example 2
receiver H2 sends reservation message for source H4 at bandwidth b
propagated upstream towards H4, reserving b
in
L1, L6
L2(H1-via-H1
out L6(H1-via-H1
L1(H4-via-R2
H2
L2
H1
in
;H4-via-R2 (b))
)
)
L4, L7
L3(H4-via-H4 ; H1-via-R2
out L4(H1-via-62 )
L7(H4-via-H4 (b))
H3
b
L1
R1
b
L6
in
R2
b
L7
L6, L7
L6(H4-via-R3 (b))
out L7(H1-via-R1 )
4
8
)
R3
L3
b
L4
H4
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RSVP: soft-state
senders periodically resend path msgs to refresh (maintain) state
receivers periodically resend resv msgs to refresh (maintain) state
path and resv msgs have TTL field, specifying refresh interval
in
L1, L6
L2(H1-via-H1
out L6(H1-via-H1
L1(H4-via-R2
H2
L2
H1
in
;H4-via-R2 (b))
)
)
L4, L7
L3(H4-via-H4 ; H1-via-R3
out L4(H1-via-62 )
L7(H4-via-H4 (b))
H3
b
L1
R1
b
L6
in
R2
b
L7
L6, L7
L6(H4-via-R3 (b))
out L7(H1-via-R1 )
4
9
)
R3
L3
b
L4
H4
Transmisión de Datos Multimedia - Master IC 2007/2008
RSVP: soft-state
suppose H4 (sender) leaves without performing teardown
eventually state in routers will timeout and disappear!
in
L1, L6
L2(H1-via-H1
out L6(H1-via-H1
L1(H4-via-R2
H2
L2
H1
in
;H4-via-R2 (b))
)
)
L4, L7
L3(H4-via-H4 ; H1-via-R3
out L4(H1-via-62 )
L7(H4-via-H4 (b))
H3
b
L1
R1
b
L6
in
R2
b
L7
L6, L7
L6(H4-via-R3 (b))
out L7(H1-via-R1 )
5
0
)
R3
L3
b
L4
gone
H4
fishing!
Tema 3:
Protocolos de transporte multimedia.
Requisitos de la red
Gestión de los recursos: IntServ vs
DiffServ
Computer Networking: A
Top Down Approach
Featuring the Internet,
RSVP
RTP/RTCP: Transporte de flujos
multimedia
RTSP: Control de sesión y localización
de medios
Protocolos para establecimiento y
gestión de sesiones multimedia
SIP
H.323
Transmisión de Datos Multimedia –
3rd edition.
Jim Kurose, Keith Ross
Addison-Wesley, July 2004.
Thanks to :
RADCOM technologies
H. Shulzrinne
Paul. E. Jones (from packetizer.com)
http://www.grc.upv.es/docencia/tdm
– Master IC 2007/2008
Transmisión de Datos Multimedia - Master IC 2007/2008
5
2
RTP (Real-time Transport Protocol)
Se basa en el concepto de sesión: la asociación entre un
conjunto de aplicaciones que se comunican usando RTP
Una sesión es identificada por:
Una dirección IP multicast
Dos puertos: Uno para los datos y otro para control
(RTCP)
Un participante (participant) puede ser una máquina o
un usuario que participa en una sesión
Cada media distinto es trasmitido usando una sesión
diferente.
Por ejemplo, si se quisiera transmitir audio y vídeo
harían falta dos sesiones separadas Esto permite
a un participante solamente ver o solamente oír
Transmisión de Datos Multimedia - Master IC 2007/2008
RTP (Real-time Transport Protocol)
Audio-conferencia con multicast y RTP
Sesión de audio: Una dirección multicast y dos puertos
Datos de audio y mensajes de control RTCP.
Existirá (al menos) una fuente de audio que enviará un stream de
segmentos de audio pequeños (20 ms) utilizando UDP.
A cada segmento se le asigna una cabecera RTP
La cabecera RTP indica el tipo de codificación (PCM, ADPCM, LPC,
etc.)
Número de secuencia y fechado de los datos.
Control de conferencia (RTCP):
Número e identificación de participantes en un instante dado.
Información acerca de cómo se recibe el audio.
Audio y Vídeo conferencia con multicast y RTP
Si se utilizan los dos medios, se debe crear una sesión RTP
independiente para cada uno de ellos.
5
3
Una dirección multicast y 2 puertos por cada sesión.
Existencia de participantes que reciban sólo uno de los medios.
Transmisión de Datos Multimedia - Master IC 2007/2008
5
4
RTP: Mezcladores y traductores
Mezcladores (Mixers).
Permiten que canales con un BW bajo puedan participar en una
conferencia. El mixer re-sincroniza los paquetes y hace todas las
conversiones necesarias para cada tipo de canal.
Traductores (Translators).
Permiten conectar sitios que no tienen acceso multicast (p.ej.
que están en una sub-red protegida por un firewall)
Transmisión de Datos Multimedia - Master IC 2007/2008
RTP: Formato de mensaje (I)
32 bits
V PX
CC M
PT
Sequence number
Timestamp
Synchronization Source (SSRC) ID
Contributing Source (CSRC) ID
V: versión; actualmente es la 2
P: si está a 1 el paquete tiene bytes de relleno (padding)
X: presencia de una extensión de la cabecera
5
5
Transmisión de Datos Multimedia - Master IC 2007/2008
RTP: Formato de mensaje (II)
32 bits
V PX
CC M
PT
Sequence number
Timestamp
Synchronization Source (SSRC) ID
Contributing Source (CSRC) ID
CC: Identifica el número de CSRC que contribuyen a los datos
M: Marca (definida según el perfil)
PT: Tipo de datos (según perfil)
5
6
Transmisión de Datos Multimedia - Master IC 2007/2008
RTP: Formato de mensaje (III)
32 bits
V PX
CC M
PT
Sequence number
Timestamp
Synchronization Source (SSRC) ID
Contributing Source (CSRC) ID
Sequence number: Establece el orden de los paquetes
Timestamp: Instante de captura del primer octeto del campo de datos
SSRC: Identifica la fuente de sincronización
CSRC: Fuentes que contribuyen
5
7
Transmisión de Datos Multimedia - Master IC 2007/2008
5
8
RTP header definition
/*
* RTP data header
*/
typedef struct {
unsigned int version:2;
unsigned int p:1;
unsigned int x:1;
unsigned int cc:4;
unsigned int m:1;
unsigned int pt:7;
u_int16 seq;
u_int32 ts;
u_int32 ssrc;
u_int32 csrc[1];
} rtp_hdr_t;
Transmisión de Datos Multimedia - Master IC 2007/2008
5
9
RTP y las aplicaciones
La especificación de
RTP para una
aplicación particular va
acompañada de:
Un perfil (profile) que
defina un conjunto de
códigos para los tipos
de datos transportados
(payload)
El formato de
transporte de cada uno
de los tipos de datos
previstos
Ej.: RFC 1890 para audio y vídeo
PT
encoding audio/video clock rate channels
name
(A/V)
(Hz)
(audio)
______________________________________________
0
PCMU
A
8000
1
1
1016
A
8000
1
2
G721
A
8000
1
3
GSM
A
8000
1
...
34-71 unassigned
?
72-76 reserved
N/A
N/A
N/A
77-95 unassigned
?
96-127 dynamic
?
PCMU
Corresponde a la recomendación CCITT/ITU-T
G.711. El audio se codifica con 8 bits por
muestra, después de una cuantificación
logarítmica. PCMU es la codificación que se
utiliza en Internet para un media de tipo
audio/basic.
Transmisión de Datos Multimedia - Master IC 2007/2008
RTCP (RTP Control Protocol)
RTCP se basa en envíos periódicos de paquetes de
control a los participantes de una sesión RTP
Permite realizar una realimentación de la calidad de
recepción de los datos (estadísticas).
Los paquetes de control siempre llevan la
identificación de la fuente RTP: CNAME
Asociar más de una sesión a un mismo fuente
(sincronización).
El envío de estos paquetes debe ser controlado por
cada participante (sistema ampliable).
Control de sesión (opcional)
Información adicional de cada participante.
Entrada y salida de participantes en las sesión.
Negociación de parámetros y formatos.
6
0
Transmisión de Datos Multimedia - Master IC 2007/2008
6
1
RTCP (RTP Control Protocol)
RTCP permite controlar el trafico que no se auto-limita
(p.ej cuando el número de fuentes aumenta)
Para ello se define el ancho de banda de la sesión. RTCP
se reserva el 5% (bwRTCP)
A cada fuente se le asigna 1/4 de bwRTCP
El intervalo entre cada paquete RTCP es > 5 sec
Transmisión de Datos Multimedia - Master IC 2007/2008
RTCP (RTP Control Protocol)
Formato de un paquete RTCP:
Existen distintos tipos de paquetes RTCP:
SR (Sender Report): Informes estadísticos de transmisión y
recepción de los elementos activos en la sesión.
RR (Receiver Report): Informes estadísticos de recepción
en los receptores.
SDES (Source Description): Información del participante
(CNAME, e-mail, etc).
BYE: Salida de la sesión.
APP: Mensajes específicos de la aplicación.
Cada paquete RTCP tiene su propio formato.
Su tamaño debe ser múltiplo de 32 bits (padding).
Se pueden concatenar varios paquetes RTCP en uno
(compound RTCP packet).
6
2
Transmisión de Datos Multimedia - Master IC 2007/2008
RTCP: Mensajes SR
32 bits
Sender
info
Report
block 1
6
3
RC
PT=SR=200
Longitud
SSRC del sender
NTP timestamp msw
NTP timestamp lsw
RTP timestamp
Contador de los paquetes del sender
cabecera V P
Contador de los bytes del sender
SSRC_1
Frac perd
Total paquetes perdidos
Num sec más alto recibido
Jitter de inter-llegada
Ultimo SR (LSR)
Retraso del último SR (LSR)
Transmisión de Datos Multimedia - Master IC 2007/2008
6
4
RTCP: Cálculo del Jitter
Es una estimación de la variancia del tiempo de interllegada de los paquetes RTP
D(i, j ) ( R j Ri ) (S j Si ) ( R j S j ) ( Ri Si )
Si RTP timestamp del paquete i
Ri Instante de llegada del paquete i
J i J i 1 Di 1, i J i 1 / 16
Tema 3:
Protocolos de transporte multimedia.
Requisitos de la red
Gestión de los recursos: IntServ vs
DiffServ
Computer Networking: A
Top Down Approach
Featuring the Internet,
RSVP
RTP/RTCP: Transporte de flujos
multimedia
RTSP: Control de sesión y localización
de medios
Protocolos para establecimiento y
gestión de sesiones multimedia
SIP
H.323
Transmisión de Datos Multimedia –
3rd edition.
Jim Kurose, Keith Ross
Addison-Wesley, July 2004.
Thanks to :
RADCOM technologies
H. Shulzrinne
Paul. E. Jones (from packetizer.com)
http://www.grc.upv.es/docencia/tdm
– Master IC 2007/2008
Transmisión de Datos Multimedia - Master IC 2007/2008
6
6
Real-Time Streaming Protocol
RFC 2326
Tiene la función de un “mando a distancia por la red”
para servidores multimedia
Permite establecer y controlar uno o más flujos de datos
sincronizados
NO existe el concepto de conexión RTSP sino de sesión
RTSP
Además, una sesión RTSP no tiene relación con ninguna
conexión especifica de nivel transporte (p.ej. TCP o
UDP)
Los flujos de datos no tienen por que utilizar RTP
Está basado en HTTP/1.1
Diferencias importantes:
No es stateless
Los clientes y servidores pueden generar peticiones
Transmisión de Datos Multimedia - Master IC 2007/2008
Terminología RTSP
Conferencia
Media stream
Una instancia única
de un medio
continuo:
Un stream audio,
Un stream vídeo
Una “whiteboard”
Voz del
conferenciante
Imagen de las
transparencias
Imagen del
conferenciante
Presentación:
Es el conjunto de
uno o más streams,
que son vistos por el
usuario como un
conjunto integrado
6
7
Imagen del
público
Voz del
público
Transmisión de Datos Multimedia - Master IC 2007/2008
RTSP: Ejemplo de una sesión
HTTP GET
Cliente
SETUP
PLAY
RTP audio
RTP vídeo
RTCP
PAUSE
TEARDOWN
6
8
Web
server
descripción de la sesión
Media
server
Transmisión de Datos Multimedia - Master IC 2007/2008
RTSP: Comandos de petición
Request =
Request-Line ;
*( general-header | request-header | entity-header )
CRLF
[ message-body ]
Request-Line = Method SP Request-URI SP RTSP-Version CRLF
Method
=
"DESCRIBE“ | "ANNOUNCE" | "GET_PARAMETER" |
"OPTIONS“
| "PAUSE" | "PLAY" | "RECORD" |
"REDIRECT" | "SETUP" | "SET_PARAMETER" |
"TEARDOWN" | extension-method
extension-method = token
Request-URI = "*" | absolute_URI
RTSP-Version = "RTSP" "/" 1*DIGIT "." 1*DIGIT
6
9
Transmisión de Datos Multimedia - Master IC 2007/2008
RTSP: Mensajes de respuesta
Response =
*(
|
|
Status-Line ;
general-header
response-header
entity-header )
CRLF
[ message-body ]
Status-Line = RTSP-version SP Status-Code SP Reason-Phrase CRLF
Status-Code =
1xx: Información (Comando recibido, procesando,..) |
2xx: Exito (Comando recibido y ejecutado con éxito) |
3xx: Re-dirección (Comando recibido pero aún no completado) |
4xx: Error del cliente (El comando tiene errores de sintaxis) |
5xx: Error del servidor (Error interno del servidor)
7
0
Transmisión de Datos Multimedia - Master IC 2007/2008
RTSP: Una sesión completa (I)
1
2
media server A
4
web server W
cliente C
3
media server V
C->W: GET /twister.sdp HTTP/1.1
Host: www.example.com
Accept: application/sdp
W->C: HTTP/1.0 200 OK
Content-Type: application/sdp
v=0
o=- 2890844526 2890842807 IN IP4 192.16.24.202
s=RTSP Session
m=audio 0 RTP/AVP 0
a=control:rtsp://audio.example.com/twister/audio.en
m=video 0 RTP/AVP 31
a=control:rtsp://video.example.com/twister/video
7
1
Transmisión de Datos Multimedia - Master IC 2007/2008
RTSP: Una sesión completa (II)
C->A: SETUP rtsp://audio.example.com/twister/audio.en RTSP/1.0
CSeq: 1
Transport: RTP/AVP/UDP;unicast;client_port=3056-3057
A->C: RTSP/1.0 200 OK
CSeq: 1
Session: 12345678
Transport: RTP/AVP/UDP;unicast;client_port=3056-3057;
server_port=5000-5001
C->V: SETUP rtsp://video.example.com/twister/video RTSP/1.0
CSeq: 1
Transport: RTP/AVP/UDP;unicast;client_port=3058-3059
V->C: RTSP/1.0 200 OK
CSeq: 1
Session: 23456789
Transport: RTP/AVP/UDP;unicast;client_port=3058-3059;
server_port=5002-5003
7
2
Transmisión de Datos Multimedia - Master IC 2007/2008
RTSP: Una sesión completa (III)
C->V: PLAY rtsp://video.example.com/twister/video RTSP/1.0
CSeq: 2
Session: 23456789
Range: smpte=0:10:00V->C: RTSP/1.0 200 OK
CSeq: 2
Session: 23456789
Range: smpte=0:10:00-0:20:00
RTP-Info: url=rtsp://video.example.com/twister/video;
seq=12312232;rtptime=78712811
C->A: PLAY rtsp://audio.example.com/twister/audio.en RTSP/1.0
CSeq: 2
Session: 12345678
Range: smpte=0:10:00A->C: RTSP/1.0 200 OK
CSeq: 2
Session: 12345678
Range: smpte=0:10:00-0:20:00
RTP-Info: url=rtsp://audio.example.com/twister/audio.en;
seq=876655;rtptime=1032181
7
3
Transmisión de Datos Multimedia - Master IC 2007/2008
RTSP: Una sesión completa (IV)
C->A: TEARDOWN rtsp://audio.example.com/twister/audio.en RTSP/1.0
CSeq: 3
Session: 12345678
A->C: RTSP/1.0 200 OK
CSeq: 3
C->V: TEARDOWN rtsp://video.example.com/twister/video RTSP/1.0
CSeq: 3
Session: 23456789
V->C: RTSP/1.0 200 OK
CSeq: 3
7
4
Tema 3:
Protocolos de transporte multimedia.
Requisitos de la red
Gestión de los recursos: IntServ vs
DiffServ
Computer Networking: A
Top Down Approach
Featuring the Internet,
RSVP
RTP/RTCP: Transporte de flujos
multimedia
RTSP: Control de sesión y localización
de medios
Protocolos para establecimiento y
gestión de sesiones multimedia
SIP
H.323
Transmisión de Datos Multimedia –
3rd edition.
Jim Kurose, Keith Ross
Addison-Wesley, July 2004.
Thanks to :
RADCOM technologies
H. Shulzrinne
Paul. E. Jones (from packetizer.com)
http://www.grc.upv.es/docencia/tdm
– Master IC 2007/2008
Transmisión de Datos Multimedia - Master IC 2007/2008
Voice-over-Data (VoD) Enables New Applications
“Click to talk” web sites for e-commerce
Digital white-board conferences
Broadcast audio and video over the Internet or a
corporate Intranet
Integrated messaging: check (or leave) voice mail over
the Internet
Instant messaging
Voicemail notifications
Stock notifications
Callback notification
Fax over IP
Etc.
7
6
Transmisión de Datos Multimedia - Master IC 2007/2008
7
7
Sesion Initiation Protocol
SIP is end-to-end, client-server session signaling protocol
SIP’s primarily provides presence and mobility
Protocol primitives: Session setup, termination, changes,...
Arbitrary services built on top of SIP, e.g.:
Redirect calls from unknown callers to secretary
Reply with a webpage if unavailable
Send a JPEG on invitation
Features:
Textual encoding (telnet, tcpdump compatible).
Programmability.
Post-dial delay: 1.5 RTT
Uses either UDP or TCP
Multicast/Unicast comm. support
Transmisión de Datos Multimedia - Master IC 2007/2008
Where’s SIP
Application
Transport
Network
Physical/Data Link
7
8
RTSP
SDP
codecs
SIP
RTP
TCP
UDP
IP
Ethernet
DNS(SRV)
Transmisión de Datos Multimedia - Master IC 2007/2008
7
9
IP SIP Phones and Adaptors
2
1
Are
true Internet
hosts
Analog phone adaptor
Choice of application
Choice of server
IP appliances
3
Implementations
3Com (3)
Columbia University
MIC WorldCom (2)
Mediatrix (1)
4
Nortel (4)
Siemens (5)
Palm
control
4
5
Transmisión de Datos Multimedia - Master IC 2007/2008
SIP Components
User Agents
UAC (user agent client): Caller application that initiates and sends SIP requests.
UAS (user agent server): Receives and responds to SIP requests on behalf of
clients; accepts, redirects or refuses calls.
Server types
Redirect Server
Accepts SIP requests, maps the address into zero or more new addresses and returns
those addresses to the client. Does not initiate SIP requests or accept calls.
Proxy Server
Contacts one or more clients or next-hop servers and passes the call requests further.
Contains UAC and UAS.
Registrar Server
A registrar is a server that accepts REGISTER requests and places the information it
receives in those requests into the location service for the domain it handles.
Location Server
Provides information about a caller's possible locations to redirect and proxy servers.
May be co-located with a SIP server.
Gateways
A Sip Gateway service allows you to call 'real' numbers from your software or
have a dedicated 'real' telephone number which comes in via VoIP
8
0
Transmisión de Datos Multimedia - Master IC 2007/2008
SIP Trapezoid
DNS
Registrar
SIP
Outgoing
Proxy
Incoming
Proxy
SIP
Originating
User Agent
8
1
Location
Server
DNS
Server
SIP
SIP
SIP
RTP
Terminating
User Agent
Transmisión de Datos Multimedia - Master IC 2007/2008
SIP Triangle?
DNS
Registrar
Incoming
Proxy
SIP
Originating
User Agent
8
2
Location
Server
DNS
Server
SIP
SIP
SIP
RTP
Terminating
User Agent
Transmisión de Datos Multimedia - Master IC 2007/2008
SIP Peer to Peer!
Originating
User Agent
8
3
SIP
RTP
Terminating
User Agent
Transmisión de Datos Multimedia - Master IC 2007/2008
8
4
SIP Methods
INVITE
Requests a session
ACK
Final response to the INVITE
OPTIONS
Ask for server capabilities
CANCEL
Cancels a pending request
BYE
Terminates a session
REGISTER
Sends user’s address to server
Transmisión de Datos Multimedia - Master IC 2007/2008
8
5
SIP Responses
1XX
Provisional
100 Trying
2XX
Successful
200 OK
3XX
Redirection
302 Moved Temporarily
4XX
Client Error
404 Not Found
5XX
Server Error
504 Server Time-out
6XX
Global Failure
603 Decline
Transmisión de Datos Multimedia - Master IC 2007/2008
SIP Flows - Basic
User
A
“Calls”
18.18.2.4
User
B
INVITE: sip:18.18.2.4
180 - Ringing
200 - OK
Answers
ACK
RTP
Talking
Hangs up
Talking
BYE
200 - OK
8
6
Rings
Transmisión de Datos Multimedia - Master IC 2007/2008
SIP INVITE
INVITE sip:e9-airport.mit.edu SIP/2.0
From: "Dennis Baron"<sip:[email protected]>;tag=1c41
To: sip:e9-airport.mit.edu
Call-Id: [email protected]
Cseq: 1 INVITE
Contact: "Dennis Baron"<sip:[email protected]>
Content-Type: application/sdp
Content-Length: 304
Accept-Language: en
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE
Supported: sip-cc, sip-cc-01, timer, replaces
User-Agent: Pingtel/2.1.11 (WinNT)
Date: Thu, 30 Sep 2004 00:28:42 GMT
Via: SIP/2.0/UDP 18.10.0.79
8
7
Transmisión de Datos Multimedia - Master IC 2007/2008
8
8
Session Description Protocol
IETF RFC 2327
“SDP is intended for describing multimedia sessions for the
purposes of session announcement, session invitation, and other
forms of multimedia session initiation.”
SDP includes:
The type of media (video, audio, etc.)
The transport protocol (RTP/UDP/IP, H.320, etc.)
The format of the media (H.261 video, MPEG video, etc.)
Information to receive those media (addresses, ports, formats and so
on)
Transmisión de Datos Multimedia - Master IC 2007/2008
SDP
v=0
o=Pingtel 5 5 IN IP4 18.10.0.79
s=phone-call
c=IN IP4 18.10.0.79
t=0 0
m=audio 8766 RTP/AVP 96 97 0 8 18 98
a=rtpmap:96 eg711u/8000/1
a=rtpmap:97 eg711a/8000/1
a=rtpmap:0 pcmu/8000/1
a=rtpmap:8 pcma/8000/1
a=rtpmap:18 g729/8000/1
a=fmtp:18 annexb=no
a=rtpmap:98 telephone-event/8000/1
8
9
Transmisión de Datos Multimedia - Master IC 2007/2008
9
0
CODECs
GIPS Enhanced G.711
8kHz sampling rate
Voice Activity Detection
Variable bit rate
G.711
8kHz sampling rate
64kbps
G.729
8kHz sampling rate
8kbps
Voice Activity Detection
Transmisión de Datos Multimedia - Master IC 2007/2008
9
1
SIP Flows - Registration
User
B
Registrar
Location
MIT.EDU
MIT.EDU
REGISTER: sip:[email protected]
401 - Unauthorized
REGISTER: (add credentials)
sip:[email protected]
Contact 18.18.2.4
200 - OK
Transmisión de Datos Multimedia - Master IC 2007/2008
SIP REGISTER
REGISTER sip:mit.edu SIP/2.0
From: "Dennis Baron"<sip:[email protected]>;tag=4561c4561
To: "Dennis Baron"<sip:[email protected]>;tag=324591026
Call-Id: 9ce902bd23b070ae0108b225b94ac7fa
Cseq: 5 REGISTER
Contact: "Dennis Baron"<sip:[email protected];LINEID=05523f7a97b54dfa3f0c0e3746d73a24>
Expires: 3600
Date: Thu, 30 Sep 2004 00:46:53 GMT
Accept-Language: en
Supported: sip-cc, sip-cc-01, timer, replaces
User-Agent: Pingtel/2.1.11 (WinNT)
Content-Length: 0
Via: SIP/2.0/UDP 18.10.0.79
9
2
Transmisión de Datos Multimedia - Master IC 2007/2008
SIP REGISTER – 401 Response
SIP/2.0 401 Unauthorized
From: "Dennis Baron"<sip:[email protected]>;tag=4561c4561
To: "Dennis Baron"<sip:[email protected]>;tag=324591026
Call-Id: 9ce902bd23b070ae0108b225b94ac7fa
Cseq: 5 REGISTER
Via: SIP/2.0/UDP 18.10.0.79
Www-Authenticate: Digest realm="mit.edu", nonce="f83234924b8ae841b9b0ae8a92dcf0b71096505216",
opaque="reg:change4"
Date: Thu, 30 Sep 2004 00:46:56 GMT
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, REGISTER, NOTIFY, SUBSCRIBE, INFO
User-Agent: Pingtel/2.2.0 (Linux)
Accept-Language: en
Supported: sip-cc-01, timer
Content-Length: 0
9
3
Transmisión de Datos Multimedia - Master IC 2007/2008
SIP REGISTER with Credentials
REGISTER sip:mit.edu SIP/2.0
From: "Dennis Baron"<sip:[email protected]>;tag=4561c4561
To: "Dennis Baron"<sip:[email protected]>;tag=324591026
Call-Id: 9ce902bd23b070ae0108b225b94ac7fa
Cseq: 6 REGISTER
Contact: "Dennis Baron"<sip:[email protected];LINEID=05523f7a97b54dfa3f0c0e3746d73a24>
Expires: 3600
Date: Thu, 30 Sep 2004 00:46:53 GMT
Accept-Language: en
Supported: sip-cc, sip-cc-01, timer, replaces
User-Agent: Pingtel/2.1.11 (WinNT)
Content-Length: 0
Authorization: DIGEST USERNAME="[email protected]", REALM="mit.edu",
NONCE="f83234924b8ae841b9b0ae8a92dcf0b71096505216", URI="sip:mit.edu",
RESPONSE="ae064221a50668eaad1ff2741fa8df7d", OPAQUE="reg:change4"
Via: SIP/2.0/UDP 18.10.0.79
9
4
Transmisión de Datos Multimedia - Master IC 2007/2008
SIP Flows – Via Proxy
Proxy
User
A
“Calls” dbaron
@MIT.EDU
User
B
MIT.EDU
INVITE: sip:[email protected]
INVITE: sip:[email protected]
100 - Trying
180 - Ringing
180 - Ringing
200 - OK
Answers
200 - OK
ACK
RTP
Talking
Hangs up
Talking
BYE
200 - OK
9
5
Rings
Transmisión de Datos Multimedia - Master IC 2007/2008
SIP Flows – Via Gateway
Proxy
User
A
“Calls” joe
@MIT.EDU
Gateway
30161
MIT.EDU
INVITE: sip:[email protected]
INVITE: sip:[email protected]
100 - Trying
Rings
180 - Ringing
180 - Ringing
Answers
200 - OK
200 - OK
ACK
ACK
RTP
Talking
Hangs up
Talking
BYE
BYE
200 - OK
200 - OK
9
6
Transmisión de Datos Multimedia - Master IC 2007/2008
SIP INVITE with Record-Route
INVITE sip:[email protected] SIP/2.0
Record-Route: <sip:18.7.21.118:5080;lr;a;t=2c41;s=b07e28aa8f94660e8545313a44b9ed50>
From: \"Dennis Baron\"<sip:[email protected]>;tag=2c41
To: sip:[email protected]
Call-Id: [email protected]
Cseq: 1 INVITE
Contact: \"Dennis Baron\"<sip:[email protected]>
Content-Type: application/sdp
Content-Length: 304
Accept-Language: en
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE
Supported: sip-cc, sip-cc-01, timer, replaces
User-Agent: Pingtel/2.1.11 (WinNT)
Date: Thu, 30 Sep 2004 00:44:30 GMT
Via: SIP/2.0/UDP 18.7.21.118:5080;branch=z9hG4bK2cf12c563cec06fd1849ff799d069cc0
Via: SIP/2.0/UDP 18.7.21.118;branch=z9hG4bKd26e44dfdc2567170d9d32a143a7f4d8
Via: SIP/2.0/UDP 18.10.0.79
Max-Forwards: 17
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SIP Standards
Just a sampling of IETF standards work…
IETF RFCs http://ietf.org/rfc.html
RFC3261
Core SIP specification – obsoletes RFC2543
RFC2327
SDP – Session Description Protocol
RFC1889
RTP - Real-time Transport Protocol
RFC2326
RTSP - Real-Time Streaming Protocol
RFC3262
SIP PRACK method – reliability for 1XX messages
RFC3263
Locating SIP servers – SRV and NAPTR
RFC3264
Offer/answer model for SDP use with SIP
Transmisión de Datos Multimedia - Master IC 2007/2008
SIP Standards (cont.)
RFC3265
SIP event notification – SUBSCRIBE and NOTIFY
RFC3266
IPv6 support in SDP
RFC3311
SIP UPDATE method – eg. changing media
RFC3325
Asserted identity in trusted networks
RFC3361
Locating outbound SIP proxy with DHCP
RFC3428
SIP extensions for Instant Messaging
RFC3515
SIP REFER method – eg. call transfer
SIMPLE
IM/Presence - http://ietf.org/ids.by.wg/simple.html
SIP authenticated identity management
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02.txt
http://www.ietf.org/internet-drafts/draft-ietf-sip-identity-
Tema 3:
Protocolos de transporte multimedia.
Requisitos de la red
Gestión de los recursos: IntServ vs
DiffServ
Computer Networking: A
Top Down Approach
Featuring the Internet,
RSVP
RTP/RTCP: Transporte de flujos
multimedia
RTSP: Control de sesión y localización
de medios
Protocolos para establecimiento y
gestión de sesiones multimedia
SIP
H.323
Transmisión de Datos Multimedia –
3rd edition.
Jim Kurose, Keith Ross
Addison-Wesley, July 2004.
Thanks to :
RADCOM technologies
H. Shulzrinne
Paul. E. Jones (from packetizer.com)
http://www.grc.upv.es/docencia/tdm
– Master IC 2007/2008
Transmisión de Datos Multimedia - Master IC 2007/2008
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Elements of an H.323 System
Terminals
Multipoint Control Units (MCUs)
Gateways
Gatekeeper
Border Elements
Referred to as
“endpoints”
Transmisión de Datos Multimedia - Master IC 2007/2008
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Terminals
Telephones
Video phones
IVR devices
Voicemail Systems
“Soft phones” (e.g., NetMeeting®)
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MCUs
Responsible for managing multipoint conferences (two or more
endpoints engaged in a conference)
The MCU contains a Multipoint Controller (MC) that manages the
call signaling and may optionally have Multipoint Processors (MPs)
to handle media mixing, switching, or other media processing
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Gateways
The Gateway is composed of a “Media Gateway Controller” (MGC)
and a “Media Gateway” (MG), which may co-exist or exist
separately
The MGC handles call signaling and other non-media-related
functions
The MG handles the media
Gateways interface H.323 to other networks, including the PSTN,
H.320 systems, and other H.323 networks (proxy)
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Gatekeeper
The Gatekeeper is an optional component in the H.323 system
which is primarily used for admission control and address resolution
The gatekeeper may allow calls to be placed directly between
endpoints or it may route the call signaling through itself to perform
functions such as follow-me/find-me and forward on busy
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Border Elements and Peer Elements
Peer Elements, which are often co-located with a Gatekeeper,
exchange addressing information and participate in call
authorization within and between administrative domains
Peer Elements may aggregate address information to reduce the
volume of routing information passed through the network
Border Elements are a special type of Peer Element that exists
between two administrative domains
Border Elements may assist in call authorization/authentication
directly between two administrative domains or via a clearinghouse
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The Protocols (cont)
H.323 is a “framework” document that describes how the various
pieces fit together
H.225.0 defines the call signaling between endpoints and the
Gatekeeper
RTP/RTCP (RFC 3550) is used to transmit media such as audio and
video over IP networks
H.225.0 Annex G and H.501 define the procedures and protocol for
communication within and between Peer Elements
H.245 is the protocol used to control establishment and closure of
media channels within the context of a call and to perform
conference control
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The Protocols (cont)
H.450.x is a series of supplementary service protocols
H.460.x is a series of version-independent extensions to the base
H.323 protocol
T.120 specifies how to do data conferencing
T.38 defines how to relay fax signals
V.150.1 defines how to relay modem signals
H.235 defines security within H.323 systems
X.680 defines the ASN.1 syntax used by the Recommendations
X.691 defines the Packed Encoding Rules (PER) used to encode
messages for transmission on the network
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Registration, Admission, and Status - RAS
Defined in H.225.0
Allows an endpoint to request authorization to place or accept a call
Allows a Gatekeeper to control access to and from devices under its
control
Allows a Gatekeeper to communicate the address of other
endpoints
Allows two Gatekeepers to easily exchange addressing information
Transmisión de Datos Multimedia - Master IC 2007/2008
Registration, Admission, and Status – RAS (cont)
T
RRQ
RCF
GK
(endpoint is registered)
ARQ
ACF
(endpoint may place call)
DRQ
(call has terminated)
DCF
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Symbol Key:
T
Terminal
GK
Gatekeeper
GW
Gateway
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Transmisión de Datos Multimedia - Master IC 2007/2008
The H323 stack
Transmisión de Datos Multimedia - Master IC 2007/2008
H323 Clients
O.S.
Client
Price
Windows
NetMeeting
+/- free
Unix (Linux)
DC-Share
nv
Sun
Sunforum
+/- free
… ...
... ...
... ...
You can find a bigger list at:
http://www.openh323.org/h323_clients.html
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IMPLEMENTACIÓN DE TELEFONÍA IP
EN UNA ORGANIZACIÓN
INTEGRACIÓN CISCO-ASTERISK
Transmisión de Datos Multimedia - Master IC 2007/2008
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CARACTERISTICAS CISCO CALL MANAGER
Solución de Telefonía IP de Cisco
Distribuible
Escalable (30000 lineas/servidor)
Soporta muchos usuarios
Sobre Windows o linux
Soporta gran variedad de teléfonos
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Transmisión de Datos Multimedia - Master IC 2007/2008
PROTOCOLOS
Sip
H323
MGCP (Megaco Protocol)
Transmisión de Datos Multimedia - Master IC 2007/2008
OBJETIVO FINAL
CAMPUS ALCOI
CISCO IP PHONE
CISCO IP PHONE
7960
1
2
ABC
7960
messages
3
directories
1
i
DEF
services
2
ABC
6
4
5
6
MNO
GHI
JKL
MNO
7
*
8
9
7
TUV
WXYZ
PQRS
0
#
*
OPER
directories
i
services
5
JKL
PQRS
messages
3
DEF
settings
4
GHI
8
9
TUV
WXYZ
0
#
settings
OPER
CAMPUS VALENCIA
CISCO IP PHONE
CISCO IP PHONE
7960
7960
1
2
3
ABC
DEF
messages
7
*
5
JKL
6
MNO
1
i
services
4
GHI
PQRS
directories
3
DEF
messages
4
GHI
9
7
WXYZ
PQRS
0
#
*
5
JKL
directories
i
services
settings
8
TUV
OPER
2
ABC
settings
6
MNO
8
9
TUV
WXYZ
0
#
OPER
GW ALCOI
CALL MANAGER
CENTRALITA
TELÉFONOS
158.42.250.141
ASTERISK
158.42.250.173
CAMPUS GANDÍA
GW KISIN
CENTRALITA
TELÉFONOS
158.42.255.237
CENTRALITA
TELÉFONOS
MD-110
GW GANDIA
CISCO IP PHONE
CISCO IP PHONE
7960
1
2
3
ABC
DEF
5
6
GHI
7
PQRS
*
1
1
JKL
MNO
7960
messages
directories
1
i
services
4
3
DEF
5
6
messages
GHI
8
9
7
WXYZ
PQRS
0
#
*
JKL
MNO
8
9
TUV
WXYZ
0
#
OPER
directories
i
services
4
TUV
OPER
2
ABC
settings
settings
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Transmisión de Datos Multimedia - Master IC 2007/2008
FUNCIONAMIENTO DE CALL MANAGER
Transmisión de Datos Multimedia - Master IC 2007/2008
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CONFIGURACIÓN CM
Interfaz Web
https://xxxxxx/CCMAdmin/Main.asp
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PARTITIONS
Dividen el conjunto de route patterns en subconjuntos de destinos
alcanzables identificados por un nombre.
Una partición contiene una lista de Route Patterns
Facilitan el enrutado de llamadas dividiendo el route plan en
subconjuntos lógicos que se pueden basar en la organización,
localización y tipo de llamada
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Transmisión de Datos Multimedia - Master IC 2007/2008
Partitions
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SEARCH SPACES
Es una lista ordenada de rutas de partición. Estas rutas se asocian a
los dispositivos (teléfonos).
Determinan las particiones que los dispositivos que hacen una
llamada buscan para que esta llamada se realice
Transmisión de Datos Multimedia - Master IC 2007/2008
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ROUTE PATTERNS
String de digitos y un conjunto de acciones
La llamada al destino se hace solo si se marca la secuencia correcta
definida en el route pattern
Se pueden usan caracteres especiales (x…) para hacer rangos, etc
Definir route patterns para diferentes tipos de llamadas: nacionales,
sin salida….
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ESQUEMA DE NUMERACIÓN
67xxx:
68xxx:
69xxx:
7xxxx:
11xxx:
Teléfonos IP HW (Vera)
SoftPhones
Teléfonos SIP
Teléfonos analógicos (fuera del Call Manager)
Teléfonos móviles
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Transmisión de Datos Multimedia - Master IC 2007/2008
Route patterns
Transmisión de Datos Multimedia - Master IC 2007/2008
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GATEWAYS
Debe haber uno por cada campus
Otro que será el router de salida general.
Coste: 3500-4000 euros
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Transmisión de Datos Multimedia - Master IC 2007/2008
Gateways
Transmisión de Datos Multimedia - Master IC 2007/2008
TRUNK CON ASTERISK
Es un enlace desde
el Call Manager
al Asterisk:
se enrutan llamadas
de uno al otro
Se define mediante
la IP del Asterisk
CAMPUS ALCOI
CISCO IP PHONE
CISCO IP PHONE
7960
1
3
DEF
4
GHI
7
*
5
JKL
directories
1
i
services
PQRS
7960
messages
2
ABC
2
3
ABC
DEF
messages
6
services
4
MNO
GHI
8
9
7
TUV
WXYZ
PQRS
0
#
*
OPER
directories
i
settings
5
JKL
settings
6
MNO
8
9
TUV
WXYZ
0
#
OPER
CAMPUS VALENCIA
CISCO IP PHONE
CISCO IP PHONE
7960
7960
1
2
3
ABC
DEF
messages
services
4
GHI
7
PQRS
*
5
JKL
6
MNO
directories
1
i
3
DEF
messages
4
GHI
9
7
WXYZ
PQRS
0
#
*
5
JKL
directories
i
services
settings
8
TUV
OPER
2
ABC
settings
6
MNO
8
9
TUV
WXYZ
0
#
OPER
GW ALCOI
CALL MANAGER
CENTRALITA
TELÉFONOS
158.42.250.141
ASTERISK
158.42.250.173
CAMPUS GANDÍA
GW KISIN
CENTRALITA
TELÉFONOS
158.42.255.237
CENTRALITA
TELÉFONOS
MD-110
GW GANDIA
CISCO IP PHONE
CISCO IP PHONE
7960
1
2
ABC
3
directories
1
i
services
1
2
7960
messages
DEF
2
ABC
3
6
4
5
6
MNO
GHI
JKL
MNO
7
*
8
9
7
TUV
WXYZ
PQRS
0
#
OPER
*
8
9
TUV
WXYZ
0
#
OPER
directories
i
services
5
JKL
PQRS
messages
DEF
settings
4
GHI
settings
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Transmisión de Datos Multimedia - Master IC 2007/2008
Trunk
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TELEFONOS
un identificador, el Device Name (3 caracteres más la dirección
MAC )
una descripción (ej . la persona asociada)
el pool al que corresponde
su estado (registrado o no)
la dirección IP del teléfono: sólo se muestra si el teléfono está
registrado
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Transmisión de Datos Multimedia - Master IC 2007/2008
Teléfonos
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Transmisión de Datos Multimedia - Master IC 2007/2008
Teléfonos II
Transmisión de Datos Multimedia - Master IC 2007/2008
Teléfonos III
Teléfono Cisco
Teléfono SIP
300 Euros
45-50 Euros
Configuración desde el CM
http://x.y.z.w:9999/
SIP_ADDITIONAL.CONF
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Teléfonos IV
[69001]
<--------- Extensión
username=69001 <--------- Podría ser el login
type=friend
record_out=Adhoc
record_in=Adhoc
qualify=no
port=5060
nat=never
mailbox=666@testmail <------ Su buzón de voz asociado (en el voicemail.conf)
host=dynamic
dtmfmode=info
context=from-internal
canreinvite=no
callerid=device <69001>
language=es
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Transmisión de Datos Multimedia - Master IC 2007/2008
Teléfonos V
Softphone Cisco
IP Communicator
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ASTERISK
funcionalidades similares a Call Manager
Soporta SIP, H.323, MGCP, IAX
Se obtiene de : ftp:/ftp.digium.com
Integra casi todos los codecs de audio
Soporte de Telefonía Tradicional
Soporte de Telefonía por Voz IP
APIs para desarrollo de nuevos servicios y aplicaciones
Integración con Bases de Datos
Integración con Aplicaciones ya desarrolladas
Código Abierto: sw libre
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Transmisión de Datos Multimedia - Master IC 2007/2008
CONFIGURACIÓN I
http://asterisk.cc.upv.es
Transmisión de Datos Multimedia - Master IC 2007/2008
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CONFIGURACIÓN II
Editar directamente ficheros *.conf
indications.conf
extensions.conf
agents.conf
queues.conf
sip.conf
voicemail.conf
asterisk.conf ……….