VoIP_H323_RTP - 08.pdf - Phare - LIP6

Master Réseaux – PTEL 2008/2009
Voice over IP:
H323 & RTP/RTCP
Rami Langar
PHARE research Team
rami.langar@lip6.fr
15 Décembre, 2008
1

Plan
H.323
• Overview : VoIP
• H.323 Entities
• H.323 Signaling models
• H.323 Call processing steps
RTP/RTCP
• Overview : UDP, TCP
• RTP & Multimedia applications
• RTCP objectives
2

What is VoIP?
VoIP (Voice over Internet Protocol), sometimes
referred to as Internet telephony, is a method of
digitizing voice, encapsulating the digitized voice into
packets and transmitting those packets over a
packet switched IP network.
Categories of VoIP :
• Private and Desktop: Skype, Microsoft Netmeeting,
gphone, Marratech, etc.
• Enterprise : Small IP phone deployments, IP PBX,
Cisco Callmanager.
• ISP/Carrier : Toll Bypass services, VOIP wholesale,
using equipment by vendors such as Cisco, Lucent,
Avaya, etc.
3

Voice over IP - the basics
Most implementations use H.323 protocol
• An ITU-T standard for bi-directional
exchange of voice, video, and data
• Applies to an IP network
• H.323 is a set of standards for group
communication
• Uses TCP for call setup & control
• Traffic is actually carried on RTP (Real Time
Protocol) which runs on top of UDP.
4

VoIP Protocols
H.323 Multimedia Standard
• H.225 RAS - Registration, Admission,
Status
• Q.931 - Call Signaling (Setup &
Termination)
• H.245 – Media flow Control (Preferences,
Capability exchange, etc.)
• Lots of G.7XX CODECS for audio
SIP – Session Initialization Protocol
• Covered in earlier course
5

Here’s how it stacks up:
H.323 Multimedia Protocol
H.225 Call setup & control – RAS (Q.931)
H.235 Security & Authentication
H.245 Call negotiation, capability exchange
H.450 Other supplemental Services
H.246 Circuit Switched Network Interop.
H.332 Conferencing
H.26X Video CODECS
G.7XX Audio CODECS
6

How they fit in: The ISO Model
ISO Model
Layer
Presentation
Session
Transport
Network
Data Link
Protocol or Standard
Applications / CODECS
H.323 & SIP
RTP / UDP / TCP
IP – Non QOS
ATM, FR, PPP, Ethernet
7

H323 – Protocol Stack
8

Comparison of Packet vs. Circuit Switching
Circuit
Call Setup Database /
Communications
Channel
SS 7 Overlay
Dedicated
Packet
H.323 & SIP
Shared
Addressing NANP IPv4 & IPv6
9

H.323 Entities
Terminal
Terminal
Gatekeeper
Réseaux IP
MCU
Gateway
RTC
RNIS
10

H.323 Equipment
Gateway
• Device that connects H.323 voice network to
non-H.323 voice network (SIP or PSTN)
• Allows H.323 terminals to communication with
non-H.323 terminals
Gatekeeper
• Provides address translation (H.323 & E.164 to
IP)
• Admission control for H.323 terminals and
gateways
• Manage bandwidth allocation
• Other optional services
11

H.323 Equipment
MCU - Multipoint Control Unit
• Endpoint that supports conferences between 3 or more
endpoints
• Can be stand-alone device (e.g., PC) or integrated into a
gateway, gatekeeper or terminal
• Typically consists of multi-point controller (MC) and
multi-point processor (MP)
MC : Multipoint Controller - handles control and
signalling for conference support
MP : Multipoint Processor - receives streams from
endpoints, processes them, and returns them to the
endpoints in the conference
12

H.323 Equipment
An endpoint that
supports 2-way
streaming with
another H.323
terminal or gateway
Originates and
terminates calls
Includes
videoconferencing
stations, hard
phones, & soft
phones
13

Signaling Models
Determines which protocol messages pass
through the gatekeeper, and which pass directly
between the two endpoints
The more messages that are routed between the
gatekeeper, the more the load and responsibility
(more information and more control)
The gatekeeper ultimately decides on the
signalling model
Media stream (RTP) never passes through the
gatekeeper function

Direct Endpoint Call Signalling

Gatekeeper Routed Call Signalling (Q.931/H.245)
16

H.225 RAS - Discovery
Registration, Admission and Status (RAS), defined
in ITU-T H.225
RAS is responsible for registration, admission, and
disengaging procedures between H.323 endpoint
and the Gatekeeper of H.323 zone they are
assigned to.
Gatekeeper Discovery: GRQ, GCF, GRJ.
• Unicast Discovery. Endpoint knows GK IP
address & register directly using UDP port 1719
• Multicast using UDP multicast address
224.0.1.41 with UDP port 1718
17

Automatic Gatekeeper Discovery
18

H.225 RAS - Registration
Registration with Gatekeeper using its
alias or E.164 address
• RRQ Registration Request using UDP port
1719 : contains transport address to be
used for the call setup.
• RCF Registration Confirm : includes
terminal ID.
• RRJ Registration Reject
• URQ Un-registration Request
• URF Un-registration Confirm
19

Endpoint Registration
20

H.225 RAS – Call admission
ARQ – admission request
• Terminal A requests admission to make a call.
• Message sent to Gatekeeper :
Seq. num, Terminal ID, Dest. Info, CRV, BW
estimation.
ACF – admission confirm
• Mode used for the call (routed/direct).
• Gatekeeper responds with IP address of the
destination terminal or the destination gateway
(if terminal is analog).
• Authorized BW.
21

Endpoint Admission To Network
22

H.323 – Call setup (Q.931)
First TCP connection: Established between
the caller and called user using the port
1720 : Call signaling channel.
• Setup
Indicates H.323 party wants to setup a
connection to called party
• Alerting
Called user has been alerted, (phone is ringing)
• Connect
Acceptance of call by called party specifying the
H.245 call control channel for capabilities
exchange
• Release complete
H.225 (Q.931) call has been released, signaling
channel is now closed.
23

H.323 – H.245 channel control
Caller establishes logical channels for
transmission of H.323 data using a second
TCP connection
Negotiates:
• channel usage
• master/slave configuration
• flow control
• Codec used.
Messages follow the recommendation H.245
using TCP ports:
• 1024-5000 ports in Cisco implementation
24

H.323 – H.245 Messages
Terminal Capability Set message
• Contains information on a terminal’s ability to send
and receive multimedia streams
• If conferencing, determination of MCU is
negotiated during this phase
• After capabilities exchange, clients have a
compatible method for transmitting media
streams; multimedia communication channels can
be opened
Master/Slave Determination
• Determines which terminal will be master which
will be slave in the call
25

H.323 – H.245 Channel Control
Terminal A
Alias:
A@domain1.com
Canal de Controle
H.245
TCP
H.245: Terminal Capability Set
Multiplex capability
Capability Table:
H.261 video
G.711, G.729
Terminal Capability Set
Terminal Capability Set Ack
Terminal Capability Set
…Ack
Terminal B
Alias:
B@domain2.com
Canal de Controle
H.245
TCP 8741
H.245: Terminal Capability Set
Multiplex capability
Capability Table:
H.261 video
G.711, Alaw64k
26

H.323 – H.245 Channel Control
Terminal A
Alias:
A@domain1.com
H.245: OpenLogicalChannel
LogicalChannel 1,
RTCP RR port 7771
Session number, RTP PT, …
OpenLogicalChannel
Terminal B
Alias:
B@domain2.com
Canal de Controle
H.245
TCP
OpenLogicalChannel
…Ack
…Ack
Canal de Controle
H.245
TCP 8741
H.245: OpenLogicalChannelAck
LogicalChannel 1,
RTCP SR port 9345,
RTP port 9344
27

H.323 – Conversation
Terminal A
Alias:
A@domain1.com
Terminal B
Alias:
B@domain2.com
Canaux audio
RTP : UDP
RTCP : UDP 7771
RTCP : UDP
Flux RTP de A vers B
RTCP RR
RTCP SR
Canaux audio
RTP : UDP 9344
RTCP : UDP
RTCP : UDP 9345
28

Call termination
Client A completes transmission of media and
closes logical channels used to transmit media
• Close Logical Channel message initiated (H.245).
• Client A transmits End Session Command (H.245)
• Client B closes media logical channels and transmits
End Session Command
• Client A closes H.245 control channel
• If call signaling channel is still open, a Release
Complete message (H.225) is sent between clients
to close this channel
• Disengage Request (DRQ) and disengage confirm
(DCF) between terminal – GK, and Gateway – GK.
29

Call termination
Terminal GK Gateway
2
3
1
2
3
1 : Release Complete
2 : DRQ
3 : DCF
30

H.323 Model
31

RTP/RTCP details
32

Application taxonomy
Elastic applications
A user waits for info from a
server.
Small end-to-end delay
preferred (non
fundamental)
Instantaneous bit rate
required: low
Packet losses recovered by
the transport protocol
through retransmission
• End-to-end delay
increases
Multimedia applications
Two users interact (in real
time)
Low delay fundamental (a
delayed packet is
equivalent to a lost packet)
Required bit-rate may be
significant
May be robust to (limited)
packet losses
33

Applications and protocol stack
34

UDP protocol
UDP (User Datagram Protocol ) permits application to
application (host to host) communication through
datagram transmission
UDP provides a layer 4 service:
• Connectionless (out of sequence packets)
• Unreliable (packet lost)
• Low overhead
• Optional checksum
Application identification through :
• Source IP address, destination IP address, source UDP
port, destination UDP port
• No rate control
No flow control (possible receiver saturation)
No congestion control
35

UDP packet format
36

TCP protocol
TCP (Transmission Control Protocol ) is the
other Internet layer 4 protocol
Main characteristics
• Connection-oriented
• Reliable and in sequence delivery
Retransmission
• Rate control
Congestion control to avoid network saturation
• unpredictable delay characteristics
37

TCP packet header
38

Using TCP for multimedia?
TCP is reliable, but
• Retransmissions cause delays
TCP is rate controlled to avoid receiver and
network congestion, but
• The available bit rate for the multimedia application
is highly variable
TCP does not support multicast
TCP cannot be used for real-time multimedia
• Non real time multimedia can be treated as file
transfer
39

UDP for multimedia
UDP supports multicast, is not rate
controlled and does not use
retransmissions, but
• Does not guarantees in-order delivery
• Does not detect and deals with packet
losses
• Does not compensate for delay fluctuations
• Does not recognize multimedia contents
IETF proposal: add RTP protocol over
UDP
40

Delay Fluctuation : TDM vs. statistical multiplex
Multiplexeur
statistique
Optimisation de la
bande passante
Introduction de gigue
et de délai
41

Example of multimedia application
42

RTP : Real Time Transport Protocol
Defined in RFC 1889
• a framework to build multimedia applications
Provides some basic mechanisms for multimedia data
transfer
It is not and independent protocol, but must be included
in the application
Composed by two different (although related) protocols:
• RTP: deals with multimedia data transport (even UDP
ports)
• RTCP: provides control and monitoring services (odd UDP
ports)
Feedback on packet delay and losses
Request/response to coding modifications
Helps in the management of the list of participants
43

RTP + RTCP
Functions available
• Payload identification (coding)
• Sequence number management
• Timestamp management
• Monitoring and performance analysis
• Participants identification
Functions NOT available
• No QoS support
No reliability, bit rate or delay guarantees
• No guarantee on correct and in order delivery
Exploits UDP checksum to detect errors
44

RTP : example of voice transmission
Consider a multi-user voice conference over RTP
Basic elements:
• One IP multicast address
• Two UDP ports
One even port for RTP, the next one (odd) for RTCP
RTP does not specify how to choose these numbers
Voice is produced according to a proper coding
technique
Voice samples are grouped in packets
• The number of samples in each packet depends on
the RTP configuration
45

RTP : example of voice transmission
Packets size should be small to keep under control the
packetization delay (should be smaller than few tens of ms)
• Few samples in each packet
Samples are encapsulated within an IP + UDP + RTP
headers
Through RTCP, each participants sends (in multicast)
statistical data
• It is possible to analyze service performance Code rate
adaptation may be envisioned to adapt the transmission
to the measured quality
IP Header
UDP Header
RTP Header
Samples
20 Bytes 8 Bytes 12 Bytes
46

RTP : example of audio-video conference
The standard suggests to use two
independent RTP flows
• Advantage: a user may access only one
of the two services
It is mandatory to synchronize the
two flows:
• The RTP timestamps of the two flows
may be used with this goal
47

RTP packet format
48

RTCP Objectives
Quality of service and congestion control
• RTCP packets are used as “low frequency” ACKs to
signal the reception quality
• Control protocol for RTP data flow
• On the basis of RTCP “report”, the server may adapt
the coding to the communication status
Estimate the number of participants to the
multicast session
• Needed to control the transmission bit rate required
by RTCP control signals which would increase too
much if the number of participants increases too
much
49

RTCP packet
Several types of RTCP packets:
• SR (Sender Report) sent by all active sources to all
participants ; includes Amount of data sent from
RTP session start-up
Total number of RTP packets sent
Total number of byte sent
• RR (Receiver Report): sent by all receivers to all
participants; sent to inform senders on the quality
of the RTP session as seen by receivers
• SDES (Source DEScriptor): source description
through a unique identifier (to identify themselves)
• BYE: session ends or one participant leaves the
session
• APP: application specific functions
50

RTCP : Report transmission speed
Four rules must be followed to generate
RTCP packets
• The RTCP traffic should be limited to a given
percentage of the data traffic (5%)
• 25% of the session bit rate is devoted to SR
packets, the remaining part to other packets
• An RTCP packet cannot be sent earlier than 5s
after the previous RTCP packet transmission
• A variable time P should be added to the
waiting time
51

RTCP : Report transmission speed
52