doc.: IEEE 802.11-03/157r0 Submission March 2003 Allen Hollister, PlantronicsSlide 1 802.11 & VoIP...

March 2003

Allen Hollister, Plantronics

Slide 1

doc.: IEEE 802.11-03/157r0

Submission

802.11 & VoIP

Allen HollisterPlantronics

March 2003


Slide 2

doc.: IEEE 802.11-03/157r0

Submission

Router

VoIP

By

Allen Hollister

Router

Router

Router

Router

Router

Router

Router

VoIP

By

Allen Hollister

ter

Hol

lis

en

All

Vo

IP

By

Voice (10) RTP (12) UDP (8) IP (20) Ethernet (14)Trailer (4)

Or 802.11 (34)Note: To view the animation,you need PowerPoint version 2002 or later

March 2003


Slide 3

doc.: IEEE 802.11-03/157r0

Submission

Router

Router

Router

Router

Router

Router

Router

RouterterHollisenAllVoIPBy

Voice (10) RTP (12) UDP (8) IP (20) Ethernet (14)Trailer (4)

Or 802.11 (34)

VoIP

By

Allen Hollister

VoIP

By

Allen Hollister

Much of QoS comes down to making the system more “TDMA” like.We want packets moving along the same path at regular intervals

March 2003


Slide 4

doc.: IEEE 802.11-03/157r0

Submission

VoIP Layers

Total HeaderOverhead inlayers 3 through5 is 40 bytes. Bygoing to theRobust HeaderCompression,this overheaddrops to 2 bytes

Layers 3through 6 aredefined byVOIPstandards, not802.11

Frequency Hopping SpreadSpectrum

Direct Sequence SpreadSpectrum

Infrared

Header in physical layer has “sync bits” that must run at the slowest possiblespeed to ensure backward compatibility.

Overhead in this layer is 96usecs per frame. Layer 1,Physical

layer

802.11 Media Access Control Layer (Mac)This layer serves to provide reliable data delivery and controls accessto the network. For VoIP applications, this access must have priority

over data applications.This layer adds 34 bytes of header overhead.

Layer 2,Link LayerIn wiredsystem

this wouldbe

Ethernetwith 14bytes of

overhead

802.2 Logical Link Control (LLC) Not part of the 802.11 standard!

Defined bythe 802.11standard.

Everythingelse defined

by adifferent

standard orproprietary

IP (Internet Protocol) - requires 20 bytes minimum of header overheadLayer 3,Network

Layer

UPD (User Datagram Protocol) - Requires 8bytes of header overhead TCP (Transmission Control Protocol)

Layer 4,Transport

Layer

RTP (Real Time Transport Protocol)

- Requires 12 to 16 bytes ofheader overhead

RTCP

Layer 5,SessionLayer

ITU-T Rec. assume supportof G.711, G.723, G.729 40mspackets is 40 bytes for G.729

& 320 bytes for G.711SIP (Session Initiation Protocol) or

H.323, or Megaco, or proprietary

Layer 6,Presentation Layer

Applications Terminal Control & managementALayer 7,pplication

Layer

802.11i - security still in committee

• VoIP sessions consist of three separate flows over the network

– The outgoing media stream

– The incoming media stream

– The Signaling and Control stream.

• Streams use different routes and they use different protocols.

– Media Streams use RTP/UDP because voice data is very fragile. If it doesn’t get there within 150 msec, its useless!

– Signaling and Control uses TCP because it is very important that the data arrive and be accurate. The time it arrives doesn’t matter so much.

March 2003


Slide 5

doc.: IEEE 802.11-03/157r0

Submission

Connection-Oriented Protocol (TCP)

Destination Address

Destination Address continued (address of first router)

Source Address

Source Address Continued Type

Total Length (16 bits)

Flags

Header Checksum

IHLVer

Fragment Offset

Type of Service

Source Address

Destination Address (Final destination address)

Options and Padding

Identifier

Time to Live Protocol (6)

Data

Data Link Trailer

Ethernet Ver .2 Header

(14 bytes

IP Header(20 bytes)

TCP Header(min length 20

octets)

Data Link Trailer Variable length

Destination Port

Window

Acknowledgement Number

Source Port

Sequence Number

Checksum Urgent Pointer

Options and Padding

FSU P RAReservedOffset

March 2003


Slide 6

doc.: IEEE 802.11-03/157r0

Submission

TCP • A port specifies an application: for example port 23 specifies Telnet, while port 25 specifies e-

mail. – A list of all assigned port numbers can be found at: http://www.iana.org/assignments/port-numbers– A combination of a port and an IP address is a “Socket”

• Sequence number specifies the number assigned to the first byte of data in the current message. The first sequence number is random. The next sequence number will be the previous sequence number plus the number of octets in the last packet. It counts data octets!

• Acknowledgement Number: Contains the sequence number of the next byte of data the sender of the packet expects to receive

Destination Port

Window

Acknowledgement Number

Source Port

Sequence Number

Checksum Urgent Pointer

Options and Padding

FSU P RAReservedOffset

U-Urgent Data (Urgent Pointer field becomes active)

A – ACK

P – Push data needs to be promptly sent to recipient

R – Reset – Abort session, error

S – Syn

F - Finish

March 2003


Slide 7

doc.: IEEE 802.11-03/157r0

Submission

Finish Acknowledge

Acknowledge (4)

Acknowledge (2)

Acknowledge (1)

Synchronize Acknowledge

Disconnect

Data Transfer (Slow Start)

Connection

TCP StartupIP Host

AIP Host

B

Data Transfer

Data Transfer

Finish

Synchronize: Sets sequence number randomly (ex 2000)

Sync acknowledge from host B sets an acknowledgement number 1 greater than the original sequence number (ex 2001) then set its own random sequence number (ex 4000).

Host A then send an acknowledgment number back of 1 plus the sequence number from host B (4001)

Once data transmission is started, the sequence number is incremented by the number of data octets. For example if an initial sequence number is 100 and 150 octets of data are sent, then the next sequence number will be 250.

Data Transfer

Synchronize

Acknowledge

March 2003


Slide 8

doc.: IEEE 802.11-03/157r0

Submission

Complete RTP PacketDestination Address

Destination Address continued

Source Address

Source Address Continued Type

Total Length

Flags

Header Checksum

IHLVer

Voice Data – 40 msec G.729 is 40 bytes, G.711 320 bytes

Data Link Trailer

Fragment Offset

Type of Service

Source Address

Destination Address

Options and Padding

Identifier

Time to Live Protocol(17)

Destination Port

Checksum

Sequence Number

Time Stamp

Synchronization Source

Contributing Source (optional)

Source Port

Length

PTMV P CCX

Ethernet Ver .2 Header

(14 bytes

IP Header(20 bytes)

UDP Header(8 bytes)

RTP Header12-16 bytes

Data Link Trailer 4 bytes

March 2003


Slide 9

doc.: IEEE 802.11-03/157r0

Submission

QoS in Packet Audio• Delay or Latency

– Problem only with two way audio, not a problem for one way (broadcast) audio such as Real Networks.– Latency in excess of 300 msec round trip (150 msec one way) is not acceptable– People perceive latency as “rudeness” and it causes people to “step” on each other words.

• Echo – TCLw is quite important! (TIA810a)– Echo is a problem for your listener. Typically, the person you are speaking with talks, his voice is heard in your receiver, some of that voice is

then coupled to the microphone in your handset which is then transmitted back to the speaker but delayed by network latency. At certain amounts of delay and amplitude, it can make it almost impossible for the person to even speak.

• Any round trip delay >28 msec requires echo canceling or very good TCLw!– VoIP always has a round trip delay >28 msec!

• Lost Packets– More than 1% to 5% noticeable– Typical intranet <.1%, Internet <4% to 100%– Chief cause: Network congestion

• Jitter – Variation in Delay– Typical: Intranet < 50 msec, internet <200 msec to ∞– Chief Cause: Queuing, CSMA systems very unpredictable

• Multiple paths through network causing some packets to arrive sooner than others. • If 802.11 VoIP phone can’t have predictable access to network, jitter is the result.

– Adaptive jitter buffer solves jitter at expense of latency• Smart jitter buffers re-adapt during silence periods.

• Audio quality – Vocoder issues, audio bandwidth– Various vocoder standards that allow for different audio bandwidths and different quality levels traded off against bandwidth.

• G.711 64 Kbits/sec ADPCM 3 KHz bandwidth• G.729 8 Kbits/sec models voice in 10 msec chunks and sends DSP coefficients to reconstruct sound.• Others: See http://www.iana.org/assignments/rtp-parameters

March 2003


Slide 10

doc.: IEEE 802.11-03/157r0

Submission

QoS - Delay• Delay > 150 msec one way is unacceptable

– The old satellite overseas links had a one way latency of 500 ms– Typical intranet delay <50 msec,– Typical internet delay (best effort) <800 msec

• Chief cause of network latency: – Queuing in Routers– Going from 100 Mbit/sec to < 1 Mbit/sec

• Additional latency due to tradeoff of packet size vs. header overhead vs. power.– Header overhead doesn’t directly affect latency for high speed links (>1Mb/s) but does so indirectly in that

more bandwidth is used. Coupled with other traffic, a congested network results that cause delay in accessing or result in lost packets.

• 58 bytes (464 bits) of header overhead using wired Ethernet VoIP– Robust Header Compression (RHC) can reduce this to 20 bytes

• 78 bytes (624 bits) using 802.11 wireless Ethernet of header overhead in VoIP

– When power must be saved for battery life issues, the only solution is to increase the voice sample size from, for example, 10 msec to 40 msec, then pushing this accumulated data over a high speed link in something under 5 msec, and finally putting the device into a power save mode for 35 msec.

• Increasing the voice sample size from 10 msec to 40 msec directly increases latency to 40 msec. You have to accumulate the entire set o samples before it can be packetized!

• Jitter is converted to latency through the use of a “jitter buffer”. Data is accumulated in the jitter buffer for the maximum expected arrival time for the packet. This time adds directly to latency.

– If one doesn’t wait for a long enough time, packets will be lost (but latency will be less)

March 2003


Slide 11

doc.: IEEE 802.11-03/157r0

Submission

Latency – Daytime, Round Trip, Scotts Valley, CA – New York

Latency NY Daytime

0

100

200300

400

500

600

1 2 3 4 5 6 7 8 9 10 11 12 13 14 15

Hop number

msec

Distance to NY 2582 miles. Round trip Transmission time in fiber for a 2582 mile distance is 44 ms (I assumed velocity of light is 117,000 miles/sec in fiber.)

Round Trip Latency Scotts Valley to New York

Max Min Average

Daytime 534 125 143

Nighttime 325 115 118

March 2003


Slide 12

doc.: IEEE 802.11-03/157r0

Submission

Round Trip Latency

Scotts Valley to Moscow Round Trip Latency

0100200300400500600700800900

1000110012001300

1 2 3 4 5 6 7 8 9 10 11 12 13 14 15

Node Numberm

sec Maximum

Minimum

Average

Ave = 319ms,Low = 222ms

Distance from SF to London is 6052 miles – 103 ms transmission time round trip

Distance from Scotts Valley to Moscow, 7362 miles, 124 msec transmission time round trip

Scotts Valley to London Round Trip Latency

0100200300400500600700800900

1000110012001300

1 2 3 4 5 6 7 8 9 10 11 12 13 14 15

Node Number

mse

c Maximum

Minimum

Average

March 2003


Slide 13

doc.: IEEE 802.11-03/157r0

Submission

Round Trip Latency - 5 mile distance Round Trip Latency - Scotts Valley to Santa Cruz

0

50

100

150

200

250

300

350

400

1 2 3 4 5 6 7 8 9 10

Node number

Ro

un

d t

rip

(m

sec)

March 2003


Slide 14

doc.: IEEE 802.11-03/157r0

Submission

QoS Codec Standards• Codec Standards

– G.711• 64000 bits/sec (8000 samples/sec and 8 bits/sample)• ADPCM with either µ law or A law applied• A 20 msec audio sample requires 160 bytes

– G.729 (Vocoder –Linear Predictive Coding)• 8000 bits/sec• Operates on 10 msec audio samples. A 20 msec audio sample would

require two G.729 samples and a total of 20 bytes.

– Other standards, G.729B, G.723.1, G.728, G.726, G.722

• Transcoding– While it is possible to transcode between standards, it is almost never a

good idea. Audio quality is lost!– Transcoding also introduces latency

March 2003


Slide 15

doc.: IEEE 802.11-03/157r0

Submission

Required Data Rate

0

10000

20000

30000

40000

50000

60000

70000

80000

90000

100000

110000

120000

Req

uir

ed D

ata

Rat

e (b

its/

sec)

10 20 40

Audio Sample (msec)

G.711 No Header Compression - 58 byte header G.729 No Header Compression - 58 byte header

G.711 RHC 20 byte header G.729 RHC 20 byte header

March 2003


Slide 16

doc.: IEEE 802.11-03/157r0

Submission

Two VoIP Sessions Streaming Over A T1 Connection

VPVPVP

20ms, VP 78 byte

20ms, VP 78 byte

20ms, VP 78 byte

At T1 Speeds, 20 ms of G.729 (78 bytes) requires .404 ms of channel time for a single Voice Packet.

At an 8 Kb/s sample rate for G.729, 20 ms of audio data results in 50 packets/sec being sent for a total of 20.2 ms of channel time. This is 2.02% of channel capacity.

Router

March 2003


Slide 17

doc.: IEEE 802.11-03/157r0

Submission

512 Byte Data Packet Arriving At Wrong Time

VPVPVP

DP 512 byte

Router

At T1 Speeds, .4ms of channel time is required for Voice Packets

512 byte Data Packet (requiring 2.65 ms of channel time), collides with a VP. Queuing prevents the packet from being lost, but causes a time delay or “jitter”.

March 2003


Slide 18

doc.: IEEE 802.11-03/157r0

Submission

Congested Network Causing Packet Loss

20ms, VP 78 byte

VP

20ms, VP 78 byte

VP

20ms, VP 78 byte

VP

A VP collides with a another packet in a congested network. There is insufficient buffer space to prevent packet loss.

20ms, VP 78 byte

VP

20ms, VP 78 byte

VP

20ms, VP 78 byte

VPDPDPDPDPRouter

March 2003


Slide 19

doc.: IEEE 802.11-03/157r0

Submission

20 ms voice packet, No other traffic

PBX 1ms

Coder, G.729

25 msec

Switch routing 1

msec

2048 byte MTU data

packet

Wan/Lan65 msec SF/NY

78 byte voice packet .4 msec

2048 byte MTU Data packet 11 msT1 uplink

Switch routing 1

msec

Jitter buffer 40 msec

Coder decompress

3 ms

PBX 1 ms

78 byte voice packet .4 msec

2048 byte MTU Data packet 11 msec

T1 downlink

137.8 msec one way latency159.8 msec one way latency

20 ms voice packet, Data Packet ahead of VP20 ms voice packet, Data Packet ahead of VP22 ms of the 75 ms is transmission time

Estimated Latency T1 WAN uplinkEstimated Latency T1 WAN uplink

March 2003


Slide 20

doc.: IEEE 802.11-03/157r0

Submission

159 msec one way latency

No Other Traffic

PBX 1 ms

Coder, G.729

25 msec

Switch routing 1

msec

256 byte MTU data

packet


78 byte voice (20ms) packet 11 msec

256 byte MTU Data packet 37 msec56 Kbps

uplink

Switch routing


Coder decompress

3 ms

PBX 1ms

78 byte voice (20ms)packet 11 msec

256 byte MTU Data packet 37 msec

56 Kbps downlink

233 msec one way latency – Not Good!

256 byte Data Packet Ahead Of VP22 ms of the 75 ms is transmission time

Estimated Latency 56 Kbps WAN uplink

March 2003


Slide 21

doc.: IEEE 802.11-03/157r0

Submission

No Other Traffic, 11Mb/s 802.11 link

Switch routing 1

msec


98 byte voice (40ms) packet .5 msec

Switch routing 1msec


second 802 device grabs channel with 2048 byte packet 17ms

98 byte voice (40ms)packet .5 msec

T1 Uplink1.54 Mb/s

No other traffic, 1Mb/s 802.11 link

802.11 RF Link.1 msec1 msec

Packetization delay & G.729 coding delay

45.5 msec

802.11 RF Link.1 msec1 msec

Coder Decompress 5.5 ms

2048 Data Packet, 1Mb/s 802.11 link

199 msec one way latency201 msec one way latency218 msec one way latency

22 ms of the 75 ms is transmission time

Estimated Latency 802.11, 40 ms VP

118 byte RF uplink packet

If 2048 byte data packet remains in from of thevoice packet, an additional 10ms delay occursat the T1 uplink

March 2003


Slide 22

doc.: IEEE 802.11-03/157r0

Submission

Internet QoS standards• The external internet has developed a series of protocols and standards that when fully implemented,

will allow telephony grade voice over the public internet. These include:

– RSVP which guarantees bandwidth through all of the routers and prevents packet loss for voice packets (something else may lose packets if the network is bandwidth saturated, but it won’t be voice).

– Diff-Serv uses the TOS (Type Of Service) field in the Internet Protocol (IP) to give priority over other data packets. This helps jitter and latency, two major areas of QoS concern.

– MPLS (Multiprotocol Label Switching) makes a router look like a switch. The entire path for a particular voice session is reduced to the equivalent of one router “hop”. This again helps latency.

– Recently the 1Gbit and now 10 Gbit Ethernet has become available. These technologies provide sufficient bandwidth as to enable very successful competition with telephony networks

– These standards are relatively new and have not been fully rolled out. But when they are, then can make VoIP over the internet as good as a regular telephone connection.

– These protocols are being used in private networks to provide high QoS for customers of these networks.

March 2003


Slide 23

doc.: IEEE 802.11-03/157r0

Submission

RSVP Controls Queuing

VP2

100 byte voice packet

DP

200 byte data packet

VP1

Classify

VP1VP1

Dequeue

Flow Specification:

Source and destination address

Protocol

Session Identifier (port/socket)

Reserved Flow Scheduling

Works in conjunction with WFQ

Reserved queue for RSVP flow

Queues serviced by weight

VP2VP2

Session 1, guaranteed bandwidth

Session 2: Guaranteed Bandwidth

DP VP2 DP (Frag) VP2 VP1 DP (Frag) VP1

DP2 Discard!!

March 2003


Slide 24

doc.: IEEE 802.11-03/157r0

Submission

Diff-Serv Router

VP2

100 byte voice packet

DP

200 byte data packet

VP1

Classify

VP1VP2

Dequeue

DP

Flow Specification:

Classify based on TOS field

Reserved Flow Scheduling

Works in conjunction with WFQ

Queues serviced by weight

Highest priority Queue

Second highest priority Queue

DP

VP2VP2 VP1 VP1

March 2003


Slide 25

doc.: IEEE 802.11-03/157r0

Submission

ToS - QoS DefinitionsService Type Purpose Example

0 Routine Baggage

1 Priority Standby

2 Immediate Coach

3 Flash Business

4 Flash Override First

5 Critical Pilot

6 Internetwork Control President Bush

7 Network Control Bill Gates

March 2003


Slide 26

doc.: IEEE 802.11-03/157r0

Submission

Multiprotocol Label Switching - MPLS• MPLS adds a 32 bit header to the IP packet. It is used by routers

to route once and switch many times– It is much faster to switch than to route. A core router may have 100,000

entries in its routing table

– A predefined “tunnel” is created that each packet must follow. Each subsequent MPLS router reads the label and forwards it in “cut-through” mode. The router doesn’t have to read all of the IP headers, only those required to make the switch.

– QoS is achieved by setting up high priority, high speed networks for high priority traffic and slower speed parallel networks for low or medium priority traffic.

– AS far as the original packet is concerned, the routers carrying it through the MPLS network appear as a single hop.

March 2003


Slide 27

doc.: IEEE 802.11-03/157r0

Submission

Signaling And Control• H.323

– ITU Standard Ver 1 released Jan 1996 Ver 2 released 1998– Consists of other standards:

• H.225 Call Signaling– Q.931 (ISDN Standard)– RAS (Registration, Admission, Status)

• H.245 Control– Originally developed to provide conferencing based communication solutions for multimedia PC’s over LAN’s– It was not designed for IP solutions. Fixed in ver 2.– What it Does

• Phase A – Call Setup• Phase B – Initial communication and capability exchange• Phase C – Establishment of audio/video communication• Phase D – Call Services• Phase E – Call termination

• SIP– IETF Standard Released July 1999– Does the same thing as H.323– Seen as the “up and coming” protocol. Currently H.323 is more common because it has a 3 year head start.

• MeGaCo (Media Gateway Control)/H.248– Joint standard with IETF and ITU– Used to interface with the PSTN network– Works with H.323 and SIP

• MGCP– Predecessor to Megaco

• Proprietary

March 2003


Slide 28

doc.: IEEE 802.11-03/157r0

Submission

SIP vs. H.323

SIP H.323Designed by IP Networking People Designed by telecom people

“WWW mentality” -sHTTP “ISDN mentality” – Q.931, ASN.1

Uses any media (but RTP is only available) H.245 (clearly defined) RTP

UDP or TCP, but mostly UDP TCP

Unicast or Multicast Unicast only (not good for conferencing)

Proxy Gatekeeper

Simpler (200 page standard) Complex (700 page standard)

Extensibility (uses code numbers) Uses proprietary field in ASN.1 – Not Extensible

Better suited to support intelligent user devices and better suited to implement advanced features

More centrally controlled

March 2003


Slide 29

doc.: IEEE 802.11-03/157r0

Submission

Objective Of H.323 Is To Enable The Exchange Of Media Streams Between H.323 Endpoints

H323 Terminal

H323 Terminal

H323 Terminal

H323 Gateway

H323 MCU

H323 Gatekeeper

Packet Network

Scope of H.323

PSTN

March 2003


Slide 30

doc.: IEEE 802.11-03/157r0

Submission

H.323 Stack

Physical Layer (IEEE 802.3, 802.5, 802.11,etc)

Logical Link 802.3, 802.5, etc

UPD (User Datagram Protocol) - Requires 8 bytes of headeroverhead

RTCP

ITU-T Rec.assume supportof G.711, G.723,

G.728, G.729H.225 CallSignalingChannel

Terminal Control & management

H.225RAS

Channel

H.245ControlChannel

VideoApplications

VoiceApplications

H.261H.263

Management and ControlMedia Agents

TCP (TransmissionControl Protocol)

IP (Internet Protocol) - requires 20 bytes minimum of header overhead

RTP (Real Time Transport Protocol) -Requires 12 to 16 bytes of header

overhead

• H.225– Is itself a two part protocol.

• 1) A variant of ITU-T recommendation Q.931, the ISDN layer 3 specification

– Used for the establishment and teardown of connections between H.323 endpoints.

• 2) RAS (Registration, Admission, & Status)

– Used between endpoints and gatekeepers & enables the gatekeeper to manage the endpoints within its zone.

• H.245 is used to manage media streams between session participants.

– Ensures media streams sent by one participant can be received understood by another

– Operates by establishing logical channels. These channels carry the media streams between endpoints. They have a number of properties

• Media type• Bit Rate• etc

H.323 Consists of a number of H.323 Consists of a number of protocols. It specifies how these protocols. It specifies how these protocols will be used.protocols will be used.

March 2003


Slide 31

doc.: IEEE 802.11-03/157r0

Submission

RAS Functions• Gatekeeper Discovery

– Enables an endpoint to determine which gatekeeper is available to control it

• Registration– Endpoint registers with a particular gatekeeper.

• Un-registration

• Admission– Endpoint request to access network for purpose of participating in a session.

• Bandwidth is specified as part of the request• Gatekeeper may turn down the request.

• Bandwidth change

• End point location– Gatekeeper translates an alias to a network address

• Disengage

• Status– Used by gatekeeper and endpoint to inform gatekeeper about call related status such as current bandwidth

• Resource Availability– Used from gateway to gatekeeper in order to inform gatekeeper of the gateways currently available capacity, such as bandwidth.

• Non-Standard– Allows proprietary information to be passed from endpoint to gatekeeper.

March 2003


Slide 32

doc.: IEEE 802.11-03/157r0

Submission

Some Messages Supported By H.225 Q.931

• Alerting– Sent by called endpoint to indicate that the called user is being alerted.

• Call Proceeding– Optional interim response sent by the called endpoint prior to sending the connect message

• Connect– An indication that the called user has accepted a call

• Setup– An initial message used to begin call establishment

• Setup acknowledge– Optional response to setup. – Can be forwarded from a gateway in case of PSTN interworking

• Release Complete– Used to bring a call to an end

March 2003


Slide 33

doc.: IEEE 802.11-03/157r0

Submission

Session Initiation Protocol - SIP

• Performs Same Function As H.323– WorldCom networks use SIP– ATT networks use H.323

• How “net heads” want to do telephony on packet networks

• Very new standard– Because of head start, H.323 is

more common now.– SIP is now being implemented

in more new applications than H.323. Seen as “wave of the future”

IETF

SIP (Session Initiation Protocol)

SDP (Session Description Protocol)

SAP (Session Announcement Protocol)

RTSP (Real Time Streaming Protocol

March 2003


Slide 34

doc.: IEEE 802.11-03/157r0

Submission

SIP Addressing• Addresses are known as SIP URLs (Uniform Resource

Locators)– Take form of user@host which is similar to e-mail address (but they

are different)• Syntax is sip:[email protected]

– Sip deals with multimedia sessions, which could include voice.

– SIP can interwork with traditional circuit-switched networks.• SIP enables the user portion of the SIP address to be a telephone

number– sip:[email protected]– Such an address could be used to cause routing of media to a gateway that

interfaces with the PSTN– To clearly indicate the call is a telephone number, supplement the URL

with: sip:[email protected];user=phone.

March 2003


Slide 35

doc.: IEEE 802.11-03/157r0

Submission

SIP Network Entities• SIP defines two classes of network entities

– Clients• An application program that sends SIP requests• A client might be a PC with a headset attachment or a SIP phone.• Clients can be in the same platform as a server.

– SIP enable the use of proxies which can be either a client or server.

– Servers• An entity that responds to those requests.• Four types of servers

– Proxy Server– Redirect Server– User Agent Server– Registrar

– SIP is a Client-server protocol.

March 2003


Slide 36

doc.: IEEE 802.11-03/157r0

Submission

Servers

• A proxy server works like a proxy server used to access the internet from a corporate LAN– The client sends requests to the

proxy and the proxy either handles the request itself or forwards it to other servers.

– To those other servers, it appears as if the request is coming from the proxy rather than from the entity hidden behind the proxy.

– Allows call forwarding/follow me services.

Request [email protected]

Response


Response

[email protected]

[email protected]

Proxy

March 2003


Slide 37

doc.: IEEE 802.11-03/157r0

Submission

Redirect Servers

• Redirect Server– Accepts SIP requests

– Maps the destination address to zero or more new addresses

– Returns the translated address to the originator of the request

– Does not initiate any SIP requests of its own.


Ack


Response

[email protected]

[email protected]

Redirect Server

Moved, contact [email protected]

March 2003


Slide 38

doc.: IEEE 802.11-03/157r0

Submission

Session Description StructureSession Description

Session Level Information

Protocol Version

Originator and Session ID

Session Name

Session Time

Media Description1

Media Name And Transport

Connection Information

Media Description 2

Media Name And Transport

Connection Information

March 2003


Slide 39

doc.: IEEE 802.11-03/157r0

Submission

SDP Syntax• Mandatory fields (test based)

– v=(protocol version)• Also marks the start of a session description and the end of any previous session

description.

– o=(session origin or creator and session identifier)

– s=(session name) • This field is a text string that could be displayed to potential session participants.

– t=(time of session)• This field provides the start time and stop time for the session.

– m=(media)• Indicates media type, the transport port to which the data should be sent, the

transport protocol (e.g., RTP), and the media format (typically an RTP payload type. The appearance of this field also marks the boundary between session level-information and media-level information.

March 2003


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Submission

SDP types - Optional

• i=(session information)• u=(URI of description) – A web address• e=(e-mail address) - of person responsible for session• p=(phone number) - of person responsible for session• c=(connection information)• b=(bandwidth information)• r=(repeat times)• z=(time zone adjustment)• k=(encryption key)• a=(attributes)

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Submission

Subfields• Some of the fields specified in

SDP have sub fields. For example:– Media Information (m) has four

sub fields• Media type

– Audio, Video, Application, Data, or Control

• Port– Port number to which media

should be sent

• Transport• Formats

– Various types of media that can be supported

– In the case of several being supported, each is listed with the preferred sent first.

• Examples (Format)– If a system wishes to receive

voice on port 45678 and can handle speech that is coded according to G.711 mu-law, then RTP payload type 0 applies and:

• m=audio 45678 RTP/AVP 0

– If a system wishes to receive voice on port 45678 and can handle speech that is coded according to any G.728 (payload type 15) GSM (payload type 3) or G.711 mu-law (payload type 0) and it prefers G.728 then:

• m=audio 45678 RTP/AVP 15 3 0

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Submission

Example: Assume that Collins can support G.726, G.729, G.728, Boss can only support G.728. Note: Many SIP headers have been omitted for clarity.

ACK sip:[email protected] SIP/2.0CSeq:1 ACKContent-Length: 0

Conversation

Invite sip:[email protected] SIP/2.0Cseq: 1 INVITEContent-Length:230Content-Typeapplication/sdpv=0o=collins 123456 001 IN IP4 Station1.work.coms=vacationi=Discussion about time off workc=IN IP4 station1.work.comt=0 0m=audio 4444 RTP/AVP 2 4 15a=rtpmap 2 G726-32/8000a=rtpmap 4 G723/8000a=rtpmap 15 G728/8000

SIP/2.0 200 OKCseq: 1 INVITEContent-Length:161Content-Typeapplication/sdpv=0o=manager 45678 001 IN IP4 Station2.work.coms=Company vacation policyc=IN IP4 station2.work.comt=0 0m=audio 6666 RTP/AVP 15a=rtpmap 15 G728/8000

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SIP Interworking With PSTN & H.323• PSTN Interworking

– Requires a “Network Gateway” (NGW) for control interworking.• NWG interfaces with SS7 database though ISUP protocol.• Translates control and signaling between the two different networks.

– Requires a Media Gateway to convert media between the two different networks

• H.323 interworking– Internet draft titled “Interworking between a SIP/SDP network and H.323” has been

created.

– Requires the use of a gateway that looks like a SIP user agent client or user agent server on the SIP side. On the SIP side it might include a Sip registrar function and/or proxy function.

– On the H.323 side of the gateway, it could contain a gatekeeper function.

– Translation between all of the H.323 messages and SIP messages occurs in this Gateway.

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Submission

802.11 and VoIP at the Physical & Link Layers

• 802.11e has gone a long way to making VoIP compatible with 802.11. – Many VoIP products are battery operated.

• Power management in 802.11e is very important

– EDCF and HCF are critical for good QoS • Create consistent access to the network providing low jitter, and low latency

– The sidelink capability is quite helpful• Can be used to save power• Can be used to talk to another device (PDA remote dialer for example) while still having contact with the AP.

– The PDA does the signaling and call setup, the VoIP handset handles the media.

• Diff-Serv at the layer 2 is a good approximation to 802.11e.– We should work with the people who are applying Diff-Serv in the routers to make sure that the standards don’t clash

• At the physical and link layers (given the ratification of 802.11e) we have to make sure that we don’t define something that would prohibit the use of VoIP.

– Example: Someone suggested in the HTG that we could get more (payload) bits/sec by forcing a minimum size payload packet of maybe 2 to 4 KB. Doing so, would make VoIP impossible as the latency would go out of sight.

• The codec standard, G.729, can code 10 msec of audio in 10 bytes. Thus 2000 bytes would code 2 seconds worth of voice. The minimum latency would then be 2 sec. This is quite a bit over the 150 msec limit!!!

• If the committee wants to do this, they should do so with the understanding that this would be a knock-out for VoIP– Of course, for those companies that really want to do this, there will be workarounds. Get ready for a new codec standard that has 10 bytes of

audio and 1990 bytes of padding

• Get bit rate up by creating a Robust Header Compression standard

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Submission

VoIP and 802.11 At Layers Above The Link Layer

• Typically, 802 does not address layers above the link layer, but I understand this is more by convention than rule.• • If we chose to do so in this case, we would find:

– Most of VoIP is already very well covered by standards at these levels.

– However, the signaling standards, while complete in themselves, are not currently compatible with each other. SIP can’t talk to H.323 without a gateway!

• It might be fair to pick a signaling standard as “the” standard for use on 802.11 products. – This would ensure that any 802.11 VoIP phone could talk with any other 802.11 phone, but:– It would put us in the middle of “wars” about this issue– Its unlikely that the members of this committee could agree as I suspect many are on one or the other side of this issue.

– Some thought could be given to selecting a codec or set of codec’s as standard. This needs caution.

• All of the signaling standards have a very good mechanism for negotiating which codec to use.• Selecting only one would stifle innovation and prevent growth, but:• Selecting a few that all must support would ensure that communication could at least be established.

– In this case, it would be very important to not prohibit other codec's from being used.

– Finally, creating some good ways to do fast, seamless, handoffs as the VoIP 802.11 phone wanders around a building would be a good idea.

• Coordinate with the IETF and other organizations who are creating a lot of new standards in this area to ensure that we remain compatible

• Security is likely to be a big issue for this technology. The 802.11i group has gone a long way to solving this problem, but:

– It is likely in the not too distant future that complete VPN capability will become part of the operating system for portable devices. – We Should track this, and make sure that we remain compatible.

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Submission

What Can The Chip Manufacturers Do?

• Get the power down!• Support the 802.11e

standard– If you believe the PSTN

system will be around for awhile, then support HCF.

• HCF is not necessary if you not using the PSTN system. If you are, then it makes life easier for the telephone systems people since that entire network runs from a central clock.

• Support the 802.11i standard.

March 2003


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doc.: IEEE 802.11-03/157r0

Submission

Thank you

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