(session initiation protocol) technical overview

7/12/2007 Page 1

Presentation by:

Kevin M. Johnson

VP Engineering & Ops

SIP(Session Initiation Protocol)

Technical Overview

vTechnologies, Inc.

7/12/2007 Page 2vTechnologies, Inc.

Who are we?

7/12/2007 Page 3vTechnologies, Inc.

Workforce Automation Software Developer

Who are we?

7/12/2007 Page 4

Who are we?

vTechnologies, Inc.

Converged Solutions Provider

WAN

T1 Links to

Other Sites

ATL

YYZ

HNL1 HNL2

Secure

SIP

Secure

SIP

DFW-3750-1DFW-SUB-3

DFW-SUB-4

DFW-SUB-2

DFW-SUB-1

DFW-PUB

DFW-WAN-2691

DFW-UNITY1,

DFW-UNITY2

DFW-3550-3

SFO/ORD SJC/RFD

PIX

PIX

DFW-3745GWDFW-2851GW

DFW-3550-2

RDU

DFW-3550-1

SRST with

RSVP

Secure

SIP

DFW-2691GW-1

SCCP

DFW-2691GW-2

SIP

Video

DFW-EXCH2DFW-DCGC-

EXCH

DFW-TFTP-MOHDFW-MPX

RSVP

HNL

NYCSJC-RFD-WAN

PSTN

DFW-2801GW

DFW-3550-4

DFW-CUPS

7/12/2007 Page 5

Who am I?

vTechnologies, Inc.

25 Years in data and telecommunications

5 years as voting member of TIA 568 Committee[568A/B 4-Pair UTP Cabling Specification]

Founding member of IEEE “Fast EthernetWorking Group”

4 years as voting member of IEEE 802.3uCommittee [100BaseTX]

2 years as voting member of ANSI X3T9.5Committee [FDDI over Copper]

SIP ?

vTechnologies, Inc. Page 67/12/2007

VOIP BASICS

VoIP (Voice over Internet Protocol) is a genericdescription of any transmission of Voice ORVideo using “Real-Time” IP networks.

VoIP is NOT a standards specification!

VoIP has become popular as a means for service providers(AT&T, Verizon etc.) to deliver voice services primarily becauseof its more efficient use of available bandwidth.

Traditional telephone networks (ISDN) are far less efficient intheir use of available bandwidth because they reserve a channelfor each connected conversation.

In data networks, everyone can use the available bandwidth tothe maximum extent, which means that the line capacity isexploited efficiently.

7/12/2007 vTechnologies, Inc.

VoIP Basics


ISDN PRI – 1.544 Mbs/sec.

23x64K B (Bearer) Channels / 1x64K D (Delta) Channel

Supports 23 Simultaneous Full-Duplex Calls

DS1 – 1.544 Mbs/sec.

1.544 Mbs/sec. Available Bandwidth

G.729 Codec = 8K Bitstream

G.711 Codec = 64K Bitstream

Half-Duplex

VoIP Basics


VoIP Basics

Codec BR (Kbps) NEB (Kbps) MOS

G.711 64 87.2 4.1

G.729 8 31.2 3.92

G.723.1 6.4 21.9 3.56

G.726 32 55.2 3.85

BR = Bit rate

NEB = Nominal Ethernet Bandwidth (one direction)

MOS = Mean Opinion Score of audio quality where 1 is worst and 5 is best

VoIP CODECs

VoIP Basics

Common VoIP Transmission MethodologiesVoice and signaling channels are strictly separated in VoIPimplementations.

Signaling sessions are typically administered by a server whichprovides control, establishes compression and othercommunication parameters.

The voice stream is established between endpoints once thesignaling session has been negotiated.

UDP (User Datagram Protocol) is the preferred protocol for voicetransmission over IP. Because UDP does not guarantee deliveryor check for packet order, it is faster and more efficient. As well,because voice packets are time sensitive, lost packets arepreferable to delayed packets.


VoIP Basics

Common VoIP Transmission Methodologies(cont.)

VoIP transmission requires a protocol that is responsible forvoice encoding, jitter, sequence order and bandwidth usage.

RTP (Real-time Transport Protocol) was developed by the IETF(Internet Engineering Task Force) to better support applicationsthat require real-time packet delivery. RTP provides the followingservices:

Payload-type identification: Indication of what kind of content is being carried – typically voice or video.

Sequence numbering: Tracking of packets for resequencing, if needed, at the receiver-side.

Time stamping: Provides information to the receiver-side for synchronization and jitter calculations.

Delivery monitoring: Provides information related to packet loss and packet delay.



IP UDP DATARTP

IP Header(Overhead)

Voice Payload(Useful Data)

VoIP Basics

TYPICAL VoIP PACKET COMPOSITION

VoIP Basics

Quality of Service

It is a common misconception that VoIP protocols incorporatesome kind of Quality of Service mechanisms that provide forimproved packet delivery. They do not!

RTP and its associated protocols do not provide any mechanismfor guaranteeing delivery or quality of service. They simplyprovide information to other applications or mechanisms thatcan provide appropriate service quality.

Because IP networks are connectionless by nature, they areinherently unreliable (Best Effort) when compared to connection-oriented network protocols such as ISDN. Therefore, voicecommunication, which has traditionally relied on a predictable,reliable infrastructure for acceptable audio quality must haveother mechanisms in place to provide acceptable voice quality inan IP network.


VoIP Basics

Quality of Service (cont.)

QoS is a function, not of protocols, but of the underlying IPnetworking infrastructure, i.e. the switching and routingequipment on the LAN/WAN.

Quality of Service, which was established as a standard by theITU in X.902, is a traffic engineering term that refers to resourcereservation control mechanisms that can be implemented in anIP network. QoS mechanisms allow the marking of packets toprioritize their movement through the network and to allocate afixed allocation of bandwidth. The current name for this markingand prioritization is Differentiated Services or DiffServ. Atthe IP layer, DiffServe uses DSCP (Differentiated Service CodePoint) markings in the IP packet header to identify the queue towhich that packet will be assigned.



Priority Level Traffic Type

0 Best Effort

1 Background

2 Standard (Spare)

Excellent Load

(Business Critical)

Controlled Load

(Streaming Multimedia)

Video

(Interactive Media)

[Less than 100ms latency and jitter]

Voice

(Interactive Voice)

[Less than 10ms latency and jitter]

Network Control Reserved Traffic

[Lowest latency and jitter]

6

7

3

4

5

VoIP Basics

QoSPriorityLevels

Lowest Priority

Highest Priority

-

+

VoIP Basics


QoS Limitations

QoS, as it currently exists, is NOT implemented on the internet andwon’t be implemented in “Internet 2” because it can’t scale for largenetworks.

Although QoS functions well on LAN-only traffic, it only functionsone-way on internet connected endpoints.

To compensate for this limitation, inbound traffic must be“throttled” to approximate QoS functionality.


1 2ABC

3DEF

4 5JKL

6MNOGHI

7 8TUV

9WXYZPQRS

* 0OPER

#

7960CISCO IP PHONE

imessages directories

settingsservices

QoS Traffic Engineering

Throttling of HTTP/FTP/

Streaming Media, etc.

VoIP Basics


QoS Requirements

Latency: Maximum Roundtrip delay of 250ms; Maximum one-waylatency of 150ms [LAN – Local ISP – Backbone Provider]

Jitter: Maximum 2ms delay

Packet Loss:

Less than 1% packet loss for a G.711 Codec

Far less than 1% packet loss for a G.729 Codec


vTechnologies, Inc. Page 18

SIP

7/12/2007

SIP Basics

What is SIP?The IETF SIP RFC 3261 Describes SIP as:

“…an application-layer control (signaling) protocol for creating,modifying, and terminating sessions with one or more participants.These sessions include Internet telephone calls, multimediadistribution, and multimedia conferences.“

SIP does not define the underlying communications protocols,but uses existing protocol standards for transmission of the voicepayload.

SIP is a request-response protocol that functions at theapplication-layer of the OSI model (not the IP layer) and worksmuch like HTTP, even using URL-like addressing to identify theendpoint that is being called.

SIP uses UTF-8 (8-bit UCS/Unicode Transformation Format)encoding which is compatible with the ASCII character set.



SIP Basics

SIP

IP

Ethernet

SIP Basics

SIP ComponentsUA (User Agent) – SIP network terminal (SIP telephone) that contains the UAC and UAS.

UAC (User Agent Client) – Element in terminal that initiates calls.

UAS (User Agent Server) – Element in terminal that answers calls from other UACs. The UAC and UAS allow peer-to-peer calls to be made using a client-server protocol.

Proxy Server – Server that receives connection requests from the UA and transfers them to another proxy server if the particular station is not in its sphere of administration.

Redirect server – receives connection requests and sends them back to the requester including destination data instead of sending them to the calling party

Location Server – receives registration requests from the UA and updates the terminal database with them.


SIP Basics

SIP Request Messaging

INVITE – Indicates a client is being invited to participate in a callsession.

ACK – Confirms that the client has received a final response toan INVITE request.

BYE – Terminates a call and can be sent by either the caller orthe callee.

CANCEL – Cancels any pending searches but does not terminate

an already accepted call.

REGISTER – Registers the address listed in the To header fieldwith a SIP server.

OPTIONS – Queries the capabilities of servers.


SIP Basics

SIP Response Messaging: A response to a SIP requestalways consists of a 3-digit integer result code and a short textualdescription.

PROVISIONAL (1xx) Request received, continuing to process the request. A server sends a 1xx (100 Trying, 180 Ringing) response if it expects to take more than 200 ms to obtain a final response.SUCCESS (2xx) The action was successfully received, understood, and accepted (200 OK).REDIRECTION (3xx) Further action needs to be taken in order to complete the request (301/2 Moved)CLIENT_ERROR (4xx) The request contains bad syntax or cannot be fulfilled – Most common = 404 Not Found.SERVER_ERROR (5xx) The server failed to fulfill an apparently valid request (503 Service Unavailable, 513 Message Too Large).GLOBAL_ERROR (6xx) The request cannot be fulfilled at any server – rare to see 6xx errors.



SIP Basics

SIP Request/Response

Diagram

ACK (2)

ACK (2)

200 OK

INVITE (1)

SIP Stateless Proxy

sip:[email protected]

INVITE (1)

INVITE (2)

INVITE (2)

MOVED

302 ACK (1)

SIP Stateful Proxy A

INVITE (2)

sip:[email protected]

200 OK

INVITE (2)

200 OK

200 OK

ACK (2)

MEDIA (RTP) PATH

SIP Stateful Proxy B

FINAL IN-CALL SIGNALING PATH

SIP Redirect Server


SIP Basics

NAT

SIP Basics

NAT Transit and STUNThe good news is that SIP uses port 5060 and is thereforesimple to configure for NAT transit.

The bad news is that RTP uses a dynamic port range (8766 to35000), so NAT transit and firewall configuration can beproblematic for the audio payload.

A SIP endpoint that is behind NAT will not be able to respond tothe external SIP proxy with a recognized IP address because itonly knows about its LAN-side (internal) IP address.

STUN (Simple Traversal of UDP through NATs) is a client-serverprotocol that provides a method for a STUN client (on the SIPUA) to send a request to a STUN server which then reports backto the STUN client what the public IP address of the NAT routeris, and what port was opened by the NAT to allow incomingtraffic back in to the network.



SIP Basics

STUN

SERVER

ALGORITHM

Test I:

Request echo from

same address, same

port

Test II:

Request echo from

different address,

different port

No NAT:

Check for firewall

NAT detected:

Remember public IP

received?

Public IP

is link's IP?

Open Internet

"Symmetric"Firewall "Full-cone"NAT

UDP blocked

received?

Test II:

Request echo from

different address,

different port

received?

yes

yes

yes

yes

no

no

no

no

Test I (Server #2):

Request echo from

same address,

same port

Test III:

Request echo from same

address, different port

Public IP

is constant?

"Restricted cone"

NAT

"Symmetric"

NAT

"Restricted port"

NATreceived?

yes

yes

no

no


SIP Basics

Server 1

Server 2

Client

NAT

"Full Cone" NAT

All requests from the same internal IP address and port are mapped to the

same external IP address and port. An external host can send a packet to

the internal host, by sending a packet to the mapped external address.


SIP Basics

Server 1

Server 2

Client

NAT

"Restricted Cone" NAT

All requests from the same internal IP address and port are mapped to the

same external IP address and port. Unlike a full cone NAT, an external host

can send a packet to the internal host only if the internal host had previously

sent a packet to it.


SIP Basics

Server 1

Server 2

Client

NAT

"Port Restricted Cone" NAT

Port restricted cone NAT is like restricted cone NAT, but the restriction includes

port numbers. Specifically, an external host can send a packet to a particular

port on the internal host only if the internal host had previously sent a packet

from that port to the external host.


SIP Basics

Server 1

Server 2

Client

NAT

"Symmetric" NAT

All requests from the same internal IP address and port to a specific destination IP

address and port are mapped to a unique external source IP address and port. If the

same internal host sends a packet with the same source address and port to a different

destination, a different mapping is used. Only an external host that receives a packet

can send a UDP packet back to the internal host.

SIP Basics

SIP Trunking: What is it & why would I use it?

A SIP trunk is a VoIP trunk that uses SIP for session control.

It enables a pure IP connection from the enterprise to thecarrier and the carrier then provides the PSTN gateway to thepublic network.

Because the VoIP trunk is now a SIP trunk, it can support voice& data as well as video, broadband & mobile traffic.

Each SIP trunk will support one full-duplex voice session ateither G.729 or G.711 compression.

SIP trunks support DID (Direct Inward Dialing), CallerID &Location, 911, 800 inbound, LD Termination, Expanded LocalCalling Areas (subject to provider’s network presence), lower LDtermination costs etc.


SIP Basics

SIP Trunking: The Downside

Implementations by SIP carriers can vary widely because of theabundance of IETF drafts and competition between carriers andmanufacturers for early dominance.

Fax support is hit and miss because it is dependent upon carrierimplementation of ITU’s T.38 (fax over IP) or some othermethod for fax support.

QoS (?) Outbound only because we are still dealing with trafficengineering that can only be enforced on the outbound-side.


SIP

Provider

SIP

Provider

SIP

Provider

Inte

rnet

UA

Call

Contr

ol

Digital

Analog

• VoIP PBXs can act as SIP gateways that can support any type of endpoint

• Incoming calls are routed based on VoIP PBX’s incoming call routes using DIDs to user extensions,

hunt groups, auto attendants etc.

• SIP Trunks can be added one at a time as call volumes increase and can typically be turned-up in 24

to 48 hours.

H323

SIP phone

Digital phone

Analog phone

IP phone

Wireless VoIP

SIP TRUNKING


GATEWAY

Thank You

(session initiation protocol) technical overview

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