overview - indian institute of technology delhipkalra/siv864/pdf/hs-lec8.pdf · 2018. 9. 26. ·...

Slide Special Module on Media Processing and Communication

http://www.cse.iitd.ac.in/~pkalra/sil864

Overview

•  Review of last class •  Protocol stack for multimedia services •  Real-time transport protocol (RTP) •  RTP control protocol (RTCP) •  Real-time streaming protocol (RTSP) •  SIP

Multimedia applications

  Audio –  Speech (CELP – type codecs) –  Music (MP3, WAV, WMA, Real)  Video (MPEG –1, 2, 4)  Conferencing  Streaming using HTTP/TCP (MP3), or RTP/UDP (Video).

3

Multimedia, Quality of Service: What is it?

Multimedia applications: Network audio and video (`continuous media’)

network provides application with level of performance needed for application to function.

QoS

4

Protocol stacks for media streaming

Internet

Compressed Video/Audio

IPv4, IPv6

UDP

RTP Layer

Data Plane Control Plane

RTCP Layer RTSP Layer

TCP

Multimedia protocol stack

MGCP/Megaco

TCP UDP

IPv4, IPv6

H.323 SDP

SIP

RTSP RSVP RTCP

RTP

H.261, MPEG

PPP AAL3/4 AAL5 PPP

Sonet ATM Ethernet V.34

Signaling Quality of Service

Reservation Measurement

Media Transport

network

link physical

Application daem

on kernel



Jitter (contd.)

Playback buffer At time 00:00:18

At time 00:00:28

At time 00:00:38



How does Sequence number and Timestamp help ?

Audio silence example:

Solution: –  After receiving no PDUs for a while, next PDU received at the

receiver will reflect a big jump in timestamp, but have the correct next seq. no. Thus, receiver knows what happened.

–  Why might this cause problems?

send

er

rece

iver

silence

Seq no.1, Tmpst 100 Seq no.2, Tmpst 200 Seq no.3, Tmpst 300

Seq no.4, Tmpst 600 Seq no.5, Tmpst 700

•  Consider audio data –  What should the sender do during silence?

•  Not send anything

•  Receiver cannot distinguish between loss and silence



Streaming performance requirements –  Sequencing – to report PDU loss –  to report PDU reordering –  to perform out-of-order decoding

–  Time stamping and Buffering –  for play out –  for jitter and delay calculation

–  Payload type identification –  for media interpretation

–  Error concealment –covers up errors from lost PDU by using redundancy in most-adjacent-frame

–  Quality of Service (QoS) feedback – from receiver to sender for operation adjustment

–  Rate control –sender reduces sending rate adaptively to network congestion

RTP

 Communicate choice of coding   Timing for data  Synchronize different streams  Communicate loss information (UDP)  Congestion avoidance (UDP)   Frame Boundaries  User identification ([email protected])  Efficient for small audio packets

RTP Header

V=version, P=padding, X=extension, CC=contributing sources, M=marker bit for frame, PT=payload type, Timestamp for first byte, SSRC=Synchronization source - independence from Lower protocol, CSRC=Contributing Source -mixer



RTP Synchronization Source

q  synchronization source - each source of RTP PDUs

q  Identified by a unique,randomly chosen 32-bit ID (the SSRC)

q  A host generating multiple streams within a single RTP must use a different SSRC per stream

RTCP

 Real Time Transport Control Protocol  Defines profile

–  Information for interpreting RTP header  Defines format

–  How data following RTP header is interpreted  Can adapt to new applications



RTCP message types

Typically, several RTCP PDUs of different types are transmitted in a single UDP PDU



… …

Last SR (LSR)

Extended Highest sequence Number Received

Interarrival Jitter

Cumulative Number of PDU Lost Fraction Lost

SSRC_1 (SSRC of the 1st Source)

Profile-Specific Extensions

SSRC_2 (SSRC of the 2nd Source)

Delay Since Last SR (DLSR)

SSRC of Sender Length (16 bits) PT=200/201 à SR/RR RC P V

Sender Info RTP Timestamp

Sender’s PDU Count

NTP Timestamp, most significant word

NTP Timestamp, least significant word

Sender’s Octet Count

Header

Report Block 1

Report Block 2

Sender/Receiver report PDUs

Signaling protocols

  VoIP predominantly uses two: –  SIP/SDP –  H.323

Session initiation protocol (SIP)

 Developed in mmusic Group in IETF –  Proposed standard RFC2543, February 1999 –  Work began 1995 –  Part of Internet Multimedia Conferencing Suite (MBONE)   Main Functions

–  Invite users to sessions •  Find the user’s current location, match with their capabilities and

preferences in order to deliver invitation •  Carry opaque session descriptions

–  Modification of sessions –  Termination of sessions

SIP overview

SIP SDP

SIP SDP

RTP UDP

RTP UDP

LDAP

DNS RSVP RSVP

TRIP

PSTN gateway lookup Address lookup

Next-Hop May trigger

Sets up

SIP components

 User Agent Client (UAC) –  End systems –  Send SIP requests

 User Agent Server (UAS) –  Listens for call requests –  Prompts user or executes program to determine response

 User Agent –  UAC plus UAS

SIP components (cont.)

 Redirect Server –  Network server - redirects users to try other server

 Proxy Server –  Network server - a proxy request to another server can “fork”

request to multiple servers, creating a search tree

 Registrar –  Receives registrations regarding current user locations

SIP architecture

1

2 3

4 5

6 7

8

9

10 11

12

SIP Client

SIP Redirect Server

SIP Proxy SIP Proxy

SIP Client (User Agent Server)

Location Service

13

14

Request Response

Media

sip:[email protected]

berkeley.edu

columbia.edu

sip.com

sales.sip.com

[email protected]

Sip architecture (cont.)

  SIP state-less proxies (near core of network) –  used for re-direction   SIP state-ful proxies (near periphery of network)

–  Keeps track of state of call –  Useful for billing/logging, authentication

 Routing –  Registration database, external database –  TRIP

SIP architecture (cont.)

  On Startup, SIP UA Sends REGISTER to Registrar

  Registration Data Provides Addresses to Reach User

  Registration Database Forms a Dynamic Routing Database of Users

  Centralized Store is Desired for Scalability

SQL/LDAP/?

Proxy Farm

INVITE

DB

Registrar

SIP architecture (cont.)   SIP INVITE Can Contain Phone Numbers

–  sip:[email protected] –  tel:17325551212

  Do Not Correspond to Users on IP Network, but PSTN Terminals

  Call Must Be Routed to Gateway

  Gateways Often Arranged Through Peering

  Which One to Use Based on Prefixes (Domestic = gw1, Europe = gw2)

  Route Table is Mapping From Prefixes to Next Hop IP address/port/transport Plus URL Rewrite Rules

1-732 regional.com

1 longdistance.com

international.com

tel:19735551212 sip:19735551212@ longdistance.com

SIP architecture (cont.)   Inter-domain Protocol for Gateway Route Exchange –  Completed February 2000

  TRIP (Telephone Routing over IP) Supports Various Models –  Bilateral agreements –  Centralized settlements provider End

Users

Gateways

Location Server

ISP A

TRIP

Front End

ISP B

SIP call flow example USER A USER B PROXY PROXY

INVITE 407 Proxy Authenticate

ACK

INVITE INVITE

INVITE

180 Ringing 100 Trying

100 Trying

180 Ringing 180 Ringing 200 OK

200 OK 200 OK

ACK ACK

ACK BOTH WAY RTP

BYE BYE

BYE 200 OK 200 OK

200 OK

Session Description Protocol (SDP)

  Used to specify client capabilities

  Example (client can support MPEG-1 video codec, and MP3 codecs)

Session Anouncement Protocol (SAP)

 Originated around same time as SIP

 Used to announce multimedia sessions –  Announcement goes to users in a multicast group –  Announcements can contain start time of session,

duration of session, etc.

H.323

 May 1995 - H.323 work started

 June 1996 - Decided by ITU-T

  Designed for exchanging multimedia over IP networks (4 versions of H.323)

H.323 Components

 Terminals  Gatekeepers  Gateways (H.323 to H.320/H.324/POTS)  MCUs

–  Multipoint Controller (MC) –  Multipoint Processor (MP)

H.323 Gatekeeper

 Address Translation –  H.323 Alias to transport (IP) address based on terminal

registration –  “email-like” names possible –  “phone number like” names possible  Admission control

–  Permission to complete call –  Can apply bandwidth limits –  Method to control LAN traffic

H.323 Gatekeeper (cont.)

 Management of gateway –  H.320, H.324, POTS, etc.  Call Signaling

–  May route calls in order to provide supplementary services or to provide Multipoint Controller functionality

 Call Management/Reporting/Logging

H.323 MCU

 Media Distribution –  Unicast - send media to one terminal (centralized in

MP; traditional model) –  Multicast - send to each receiver directly –  Hybrid - some of each  Manage Ad Hoc multipoint calls

–  Join, invite, control of conference modes  Multiprotocol via Gateways

H.323 protocol stack

H.225.0 H.245

G.7xx H.26x

RTP

RTCP

Gate- keeper Reg, Adm, Status (RAS)

Control Data Audio Video A/V Cntl Control

TCP UDP

IP

T.120

H.323 protocol stack (cont.)

 H.323 - System Document  H.225.0 - Call Signaling, Packetization

–  Gatekeeper Registration, Admission, and Status  H.245 - Control (also used in H.324, H.310)  T.120 - Data and Conference Control

H.323 example

A Call Setup Example

 a point to point call

 One Gatekeeper using the Direct Call Model

H.323 example (cont.)

PictureTel

PictureTel

PictureTel

(1) ARQ Can I call “Bob”?

(2a) GK resolves “Bob” to IP address through H.323 registration or external name service (e.g. DNS, ULS, etc..) (2b) Admission Policy Applied (3) ACF

Yes, use this IP Address

Bill Bob

GK


PictureTel

PictureTel

PictureTel

(4) SETUP (Create)

Bill Bob

GK

(5) ARQ May I answer?

(6) ACF Yes

(7) ALERTING

(8) CONNECT (User answers)


PictureTel

PictureTel

PictureTel

Bill Bob

(9) H.245 connection established

- Capability Exchange

- Open Logical Channels (audio, video)

overview - indian institute of technology delhipkalra/siv864/pdf/hs-lec8.pdf · 2018. 9. 26. ·...

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