sip-fundamentals-voicecon-orl-2010-2-100330105339-phpapp01

8/7/2019 sip-fundamentals-voicecon-orl-2010-2-100330105339-phpapp01

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SIP FundamentalsVoiceCon Orlando 2010

Dan York, Director of Conversations, Voxeo

March 23, 2010

With thanks to David Bryan of Cogent Force


2/100

Quick Survey Who Are You?

Type of business:

Enterprise?

System integrator? reseller?

Government?

Other?

Have a SIP system today? Considering one?

SIP experience?

I know how to spell it

Worked with it a bit

Wrote several of the SIP RFCs


3/100

www.7ducattacks.com

About Dan York

blogs.voxeo.com/speakingofstandards

www.ietf.org

www.voxeo.com

www.sipforum.org

www.disruptivetelephony.com


4/100

About Voxeo

Founded in 1999

Worlds largest hosted VoiceXML and CCXML platform Over82,000 hosted ports globally; hundreds of premise deployments

Over 100,000 developers using Voxeo platforms

The Voxeo difference: Unlocked Communications, CustomerObsession Teams, Communications Passion

Services/software available for free: www.voxeo.com/free


5/100

Voxeo - Built for VoIP Since Day One

100% SIP VOIP-basedcommunications applicationplatform

Built from the ground up; nativelySIP-based

Other XML IVR platforms were builton a CT card-centric model with SIPadded afterwards as a virtual oremulated CT card

Works with the widest variety ofSIP providers, gateways,softphones and adapters

SIPMethod SIP Servlets Platform

IM, Presence, Session BorderController


6/100

Voxeo: Design Once Deploy Anywhere


7/100

A Quick Background


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Some (brief!) Background on VoIP

Packet Switched, not Circuit switched

There is no dedicated path that traffic follows

Segmented into IP packets, sent to target, particularpath not specified

Original telephone network was circuit switched(think of the operator w/wires)

SS7, defined by the ITU (International

Telecommunications Union)


9/100

Circuit Switched Network

There is a dedicated, fixed line (usually

physically connected) between parties, andreserved for their use


10/100

Packet Switched Network

There is no physical connection. The call is

chopped into packets, which are sent overa network of devices, perhaps using

different paths


11/100

Public Switched Telephone Network (PSTN)


12/100

ITSP

PSTN

InternetTelephony

ServiceProvider

ITSPITSP

ITSP

ITSPITSPITSP

ITSP

ITSP

ITSP

ITSP

ITSP

ITSP

ITSP

ITSP

ITSP

ITSP

ITSP

ITSP

ITSP

ITSP

ITSP

ITSP

ITSP ITSP

ITSP

ITSP

ITSP

ITSPITSP

ITSP

ITSP

ITSP

ITSP

Today...


13/100

PBX

VoicemailPhysical

Wiring

PSTN

Gateways

1.1Traditional Telephony


14/100

Physical

Wiring

IP

Network

IP-PBX

Voicemail

PSTN

Gateways

Mobile

Devices

IM

Networks

Web

Servers

Email

Servers

Desktop

PCs

Operating

Systems

Firewalls

Internet

Directory

Servers

VoIP

CRM

Systems

Social

Networks

Database

Servers

1.2

Application

Servers

VoIP and Unified Communications


15/100

VoIP Protocols

H.323 ITU defined VoIP protocol

ITU is very formal, government oriented

More telephony in nature than Internet

SIP IETF (Internet Engineering TaskForce) defined

protocol for general multimedia session establishment

IETF is much less formal, individual oriented

More Internet in nature than telephony


16/100

H.323 vs. SIP

Feature H.323 SIP

Encoding Binary Text

Control Centralized server Endpoints (using

proxy)Intelligence Most in central server;

core

Most in the

endpoints; edge

Modularity Monolithic (all in one

server)

Can be multiple

servers

Defined by ITU IETF

Media RTP RTP


17/100

SIP Basics


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SIP

Session Initiation Protocol

Defined (mostly) in IETF RFCs 3261-3264

Many other drafts extend SIP and add newfeatures

Text based, very similar to HTTP, mail protocols

Establishes a media sessionbetween endpoints

Allows mobility locate users using a SIP URI (sip:[email protected])

Multiple endpoints can be associated with single SIP URI

Usually runs on port 5060, using UDP or TCP

Allows for IPv4 and IPv6 (Next generation mobile phones use SIP overTCP/IPv6)


19/100

SIP

Offers Event management (Subscribe/Notify)for presence etc.

Extended by SIMPLE to support IM (Instant

Messaging)

Designed to allow easy interoperation withother Internet features such as WWW and

email


20/100

Why is SIP Important?

Its all about standards

The promise of interoperable equipment

A Nortel phone talking to a Cisco proxy toreach another user with an Avaya phone

Designed for the future

Any media (voice, video, text, ???)

Internet/endpoint centric design

Easily extensible

SIP has become the dominant VoIP

protocol


21/100

Major Components of SIP

SIP systems specify components logically

There are several components specified UA(User-Agent, Endpoint) the phone or application

Registrar keeps track of where the user is within asystem

Redirect Server used to inform devices when they needto contact different locations

Proxy Server used to relay messages back and forth

within the system

In practice, several logicalfunctions may actuallyreside in the same physicalserver/program


22/100

User Agents

A User Agent (also called a UA, Phone, orEndpoint) represents a user of a SIP system A particular user may have more than one UA

Can be hard device (fixed or mobile) or a soft client an application running on a PC

Soft ClientHard Phone Mobile Hard Phone


23/100

UAs and Proxies

Many things are really built on UAs Session Border Controllers are usually 2 UAs connected

B2BUA or Back-to-Back User Agent

(sometimes also an ALG -- special kind of firewall)

Voice mail servers

Interactive Voice Response systems

SIP Proxy Often what is sold as a proxy is really a Proxy/Registrar/

Redirect server

Proxy is often part of IP-PBX, call server, etc.

Softswitch often incorporates allof this


24/100

Proxy

Alice

3.1

BobMedia

SIPSIP

Basic SIP Call Flow


25/100

SIP

Proxy

A

Alice BobMedia (RTP, MSRP, etc.)

SIPSIP

SIP

Proxy

BSIP

SIP & Media Paths


26/100

Proxy

sip.bob-office.com

sip.bob-isp.com

I need to call Bob!

Call here for

Bob from Alice

Call for you

from Alice!

Bob is over at

bob-isp.com.Ill proxy the

call over there!


27/100

Registration

Im Alice, and Ill be using a

phone at 192.168.0.1 today!

192.168.0.1 Registrar

sip.cogentforce.com

When I get a

call for Alice, Illknow to

contact her at192.168.0.1!


28/100

SIP

Proxy

A


SIPSIP

SIP

Proxy

BSIP

SIPRegistrar

SIP1

2

Registration


29/100

Redirection

sip.cogentforce.com

sip.bob-office.com

I need to

call Bob!

Call here for

Bob from Alice

Call for youfrom Alice!

sip.bob-isp.com

Try him at

bob-isp.com

Call here

for Bobfrom Alice


30/100

SIP

Proxy

A


SIPSIP

SIP

Proxy

BSIP

The Simple Picture


31/100

Internet (or WAN)

Alice Bob

Media

SIPSIP

SIPProxy

A SIP

SIPProxy

B

SIPProxy

N

SIPProxy

D SIP

SIPProxy

CSIP SIP

Media

Proxy

A

Media

Proxy

BMedia Media

The Complex Reality


32/100

SIP Messages

SIP messages fall into two types requestsandresponses

Requests include a method telling what it is theyare doing.

REGISTER, to register where to contact a user

INVITE, to start a new session (call)

BYE, to end a session that is established

CANCEL, to end a session that is still being set up (no

final response yet) ACK, part of the triple handshake

Several others


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SIP Messages

Responses are numeric, much like HTTP

1xx Provisional 100 Trying, 180 Ringing

2xx Successful 200 OK 3xx Redirection 301 Moved Permanently, 302

Moved Temporarily

4xx Failure 404 Not Found, 410 Gone, 403Forbidden

5xx Server Failure 503 Service Unavailable

6xx Global Failure 600 Busy Everywhere


34/100

Registration Call Flow

SIP Server

REGISTER

401 Unauthorized

REGISTER

200 OK


35/100

Basic Call Flow

SIP Server

INVITEINVITE

100 Trying

180 Ringing 180 Ringing

200 OK

200 OK

ACKACK

BYEBYE

Bi-directional Media between endpoints

200 OK200 OK


36/100

No-answer Call Flow

SIP Server

INVITEINVITE

100 Trying


487 Request Terminated487 Request Terminated

ACKACK

CANCEL

200 OK

CANCEL

200 OK


37/100

AORs and Contacts

In SIP, a registrar maps between an AoR and acontact

AoR: Address ofRecord

Permanent, something on your business card Not tied to a specific device

sip:[email protected]

Contact

Ephemeral, tied to a device

sip:[email protected]

Being able to change AoRContact mapping allowsmobility


38/100

Registration, revisited

Im Alice ([email protected]), and

Ill be using a phone at 192.168.0.1 today!

Registrar

AoR Contact Map:

[email protected] [email protected]

192.168.0.1


39/100

Registration, revisited

Registrar

AoR Contact Map:

[email protected] [email protected]

10.0.1.1

Lunch time! Mobile Time!

([email protected])will be using 10.0.1.1 now!


40/100

INVITE message

INVITE sip:[email protected] SIP/2.0

Via: SIP/2.0/TCP aliceua.phonecompany.com:5060

From: Alice ;tag=ss95cbav

To: Bob

Call-ID: [email protected]

CSeq: 1 INVITEMax-Forwards: 10

Contact:

snip!...

(Alice is going to call Bob)


41/100

INVITE message


SIP request, method is INVITE, target is an AoR (for Bob)Via: SIP/2.0/TCP aliceua.phonecompany.com:5060

Via header tracks where it has been (more on this later)From: Alice ; tag=ss95cbav

Message is from Alice, and this is the SIP URL for Alice. The tag is unique

to this call and selected by Alices UA

To: Bob Alice is trying to contact Bob at his SIP URL (a tag will be selected byBobs UA later


Call-ID is a unique ID to track this particular call (selected by Alices UA)CSeq: 1 INVITE

Used to track which responses go with which messages. Responses to

this invite will also have a CSeq of 1 INVITEMax-Forwards: 10

Used in routing (more on this later)Contact:

How to contact Alice directly. Notice this is a contact (has a host)snip


42/100

Calls and Dialogs

Acallis an informal term for communicationbetween two devices

Adialogis a more specific term, which defines a

relationship between two UAs for a duration In many ways, this is what most people would think

of as a call

Defined by Call-ID, To tag, and From tag

Sender selects Call-ID, and From tag

Reciever selects To tag


43/100

Transaction

Atransactiondefines a particular operation that iscarried out between the endpoints Example: (INVITE) Set up the call, (BYE) end the call

Includes all responses (including the ACK)

May be several transactions within a dialog

Each transaction uses a new CSeq Command Sequence

Chosen by the initiator of the transaction

Each side must increase the CSeq for each newtransaction they initiate no specified starting point

But since some transactions are initiated by each side,within a call it might look out of order


44/100

Dialogs and Transactions

SIP Server

INVITEINVITE

100 Trying


200 OK

200 OK

ACKACK

BYEBYE

Bi-directional Media between endpoints

200 OK200 OK

Tra

nsaction1

Transaction2

Dialog/Call


45/100

Routing


46/100

Internet (or WAN)

Alice Bob

Media

SIPSIP

SIPProxy

ASIP

SIPProxy

B

SIPProxy

N

SIPProxy

DSIP

SIPProxy

CSIP SIP

Media

Proxy

A

Media

Proxy

BMedia Media

The Complex Reality


47/100

Routing

Each entity forwards message along path toreach destination Places next destination into Request-URI (top line)

Adds themselves as another Via

Max-Forwards is decremented to prevent loops andlimit the length of path

if it reaches 0, the message isnt forwarded again


48/100

INVITE message




To: Bob


CSeq: 1 INVITEMax-Forwards: 10

Contact:

snip!...

Lets see what happens as this messagepasses through the big company proxy on the

way to Bob


49/100

INVITE message


Via: SIP/2.0/TCP proxy.bigcompany.com:5060



To: Bob

Call-ID: [email protected]: 1 INVITE

Record-Route:

Max-Forwards: 9

Contact:

snip!...

Notice the new URI, added VIA, anddecremented Max-Forwards. This message is

forwarded to Bobs UA.


50/100

Routing Responses

When Bobs UA responds, it unrolls the Viaheaders to decide where to route the responses

The response traces the path back, sending the

response to the Via address of the entity thatsent the message (responding!)

That entity removes itself from the Via andpasses the message along, unrolling until it

reaches the sender

Notice that the proxy needs no state here


51/100

INVITE message


Via: SIP/2.0/TCP proxy.bigcompany.com:5060



To: Bob

Call-ID: [email protected]: 1 INVITE

Record-Route:

Max-Forwards: 9

Contact:

snip!...

But one more here. What is record route?


52/100

Record Route

Once a UA has the address of the other UAin the call, new messages can be sentdirectly between devices if they wish

Record route is a way for a proxy to requestkeep me in the loop Ensures that future transactions include this proxy,

rather than going directly to the other UA


53/100


54/100

Forking

While somewhat complicated, many featuresrequire forking

Send a message to more than one entity Need totrack (and manage both sides)

Parallel forking (send to more than one at a

time) Ring both desk and mobile at once, cancel the call

to whichever fails to pickup

Can also use redirection (sequential)

Ring desk phone, then try mobile if no answer

Can combine

Ring desk phone and mobile, then VM if no answer


55/100

Parallel Forking

SIP Server Desk Phone

Caller

Mobile

time a

time a

Two devices are tried at the sametime, time a


56/100

Sequential Forking

SIP Server Desk Phone

Caller

Voice Mail

time a

time b

One device is tried at time a, thenanother at time b


57/100

Parallel & Sequential Forking

SIP Server

Desk Phone

Caller

Voice Mail

time a

time b

Several devices are tried inparallel at time a, then forked to

another at time b

Mobile Phone

time a


58/100

SIP Trunking & PSTN Interconnection


59/100

Gateways

How about connecting to the outside world? If you want to talk to another SIP user, your proxy

may know how to contact that users proxy directly,and the call is all SIP

To call out to a regular phone (or to have calls comein) you use a gateway. It connects to one or moreregular phone lines to let you receive and place callsto old fashioned Plain Old TelephoneService (POTS)

Can also gateway between SIP and other protocolssuch as H.323, or to networks such as Skype


60/100


61/100

SIP Trunking

Instead of using a local gateway, use SIP toconnect to a remote gateway at a:

Internet Telephony Service Provider (ITSP)

SIP Service Provider

(other names)

Create a SIP Trunk out to remote system

NOT a physical connection like a TDM trunk. Purely a

virtual connection from your premise system to the IPaddress of the ITSP.

Can be over public Internet or over WAN, dedicatedlines or VPN


62/100

PBX

Corp HQ

Carrier PSTN

Internet

Trunking before SIP


63/100

PBX

Corp HQ

ITSP

Internet

PSTN

SIP Trunking


64/100

PBX

Corp HQ

PSTN

ITSP 1

InternetITSP 2

ITSP 3

SIP Trunking Multiple ITSPs


65/100

PBX

Corp HQ

ITSP

Internet PSTN

PBX

Office

A

SIP Trunking - Failover


66/100

PBX

Corp HQ

PSTN

ITSP(Boston)

InternetITSP(Paris)

ITSP(Tokyo)

SIP Trunking - Geography


67/100


68/100

The Many Protocols Used In SIP


69/100

Media (RTP/RTCP)

The (Many) Protocols in a SIP Call

SIP ServerSIP/SDP SIP/SDP

UA UA


70/100


71/100

Codecs (encoding)

The media that flows using RTP is encodedusing a codec

Codec is about how the audio has beenconverted to packets Example: g.711, g.729, GIPS

In general, higher quality = more network bandwidth

Codec to be used, and any parameters to be used aredescribed in SDP


72/100

SDP/RTP

Both of these are older IETF protocols used bySIP IETF tries to promote reuse whenever possible

Session Description Protocol Used by SIP to describe the media format, encoding,

destination, etc.

Included in SIP message as a payload

Real-time Transport Protocol Used to transport the encoded media across the wire

Can be used for media other than voice


73/100

INVITE message


Via: SIP/2.0/TCP aliceua.phonecompany.com:5060From: Alice ; tag=ss95cbav

To: Bob Call-ID: [email protected]

CSeq: 1 INVITE

Contact: Content-Type: application/sdp

Content-Length: 134

v=0

o=CogentForce-UA 10010 605 IN IP4 aliceua.phonecompany.coms=SIP Call

c=IN IP4 192.0.1.1t=0 0

m=audio 49172 RTP/AVP 0 8 16


74/100

INVITE message

Content-Type: application/sdpContent-Length: 134

The content of the message is SDP to describe the media of thecall. We have 134 bytes of SDP

v=0

v is theSDP version number -- 0o=CogentForce-UA 10010 605 IN IP4 aliceua.phonecompany.com

o is a description of the UA and a session IDs=SIP Call

s is an string to encode a name for this session

c=IN IP4 192.0.1.1c tells the other side what address to send the media to

t=0 0t indicates start and end times. 0 and 0 dont specify times in advance

m=audio 49172 RTP/AVP 0 8 16

m describes the media. We are listening for audio of typeRTP/AVP on port 49172


75/100

Offer/Answer

Different UAs may support different codecs

Calling UA lists what it supports and is willing touse in the INVITE message SDP

Called UA responds with the subset of that it iswilling to use in the response SDP (200 OK)

If nothing in common, cant communicate andcall is rejected

Mechanisms exist to change this mid call (forexample, switching from audio to video)


76/100

Instant Messaging / SIMPLE


77/100

Instant Message and Presence

SIP has a set of extensions calledSIMPLE SIMPLE: SIP for Instant Messaging

(IM) and Presence LeveragingExtensions

Send text messages between devices

Can switch between text and voice

session is all SIP

Presence (ability to see status of

other user) is also supported (andpersistent on server)

Idle, Offline, Available, Busy, Offlinetill Monday


78/100

Instant Message and Presence

MESSAGE method used to send textmessages

SUBSCRIBE to express an interest,

subscribe to someones status

PUBLISH to change your status on thepresence server

NOTIFY is send from server to those whohave subscribed


79/100

Extending SIP

SIMPLE isnt the only extension

SIP was designed from the beginning to be highly extensible

Can add new methods, new responses, headers

Groups in the RAI (Realtime Applications and Infrastructure)area at IETF determine what is really an accepted standardextension

SIPCORE, DISPATCH, SIMPLE, and others

Many vendors add non-interoperable extensions to SIP

These may not work across different vendor platforms


80/100

NAT / Firewall Traversal

NAT/FW T l


81/100

NAT/FW Traversal

SIP has traditionally had some trouble gettingthrough firewalls (FWs) and NetworkAddressTranslators (NATs)

Two major causes NATs use private IP addresses, not routable fromthe outside

SIP embeds these, outside entity has no way to respond

SIP and RTP (the media) are on different ports

Each requires a specific forwarding through the firewall towork and must be opened separately

Fi ll d M di


82/100

Firewalls and Media

A UA Contactsanother UA (or a

proxy), sending SIP

messages.

The process of

sending themessage (on most

NATs/FWs) creates

a mapping that

allows responses to

reach the senderSIP

Fi ll d M di


83/100

Firewalls and Media

Media (RTP) is on adifferent port

(different opening in

the FW) than SIP

It is possible that at

the time the mediais sent, it is rejected

by the firewall

RTP

NAT/FW T l


84/100

NAT/FW Traversal

Several Solutions IETF efforts

STUN, TURN and ICE

Query outside servers to find out public address

Can use media relays to help get media between two hostsbehind firewalls

Session Border Controllers (SBCs)

Sit near (or in) the firewall/NAT, rewrite the internals of thepackets, work with firewall to open ports

S i B d C t ll


85/100

Session Border Controllers

Implemented either as special firewall/ALG thatmodify the packets (deep-packet inspection) or as aB2BUA

B2BUA can be thought of as a special SIP-to-SIP gatewayor as two UAs glued together, where each side is aseparate call

Can be used for

NAT Traversal

Protocol correction

Demarcation point

Codec translation

Generally handles both media and signalling

S i B d C t ll


86/100

Session Border Controller

SIP Server

SIP/IP network

SIP Server

SIP/IP network

Session Border Controller

(SBC)


87/100

Peer-To-Peer SIP

P t P (P2P) SIP


88/100

Peer-to-Peer (P2P) SIP

Alternate way to deliver SIP

Basic idea reduce or completely eliminatethe central servers (proxy/softswitch)

Most or all functionality is moved to the UA

Highly scalable, so good for very smalldeployments (no infrastructure) and

internet-wide deployments

Cli t/S S i


89/100

Client/Server Session

In a Client/Server session, two nodesmust use a central server to

communicate

P2P Session


90/100

P2P Session

In a Peer-to-Peer session, when twonodes communicate, a few other nodes,

rather than a central server, helpcomplete the call


91/100

Distributed Systems

Home Teleworkers


92/100

7.1

UCSystem

Corp HQ

InternetFirewall HomeFirewall

IPPhone

PC

Home

Home Teleworkers

Mobile Workers


93/100

UCSystem

Corp HQ

InternetFirewallWiFiCaf

Router

Mobile

UCclient

LaptopUC

client

Mobile

DataNetwork

Mobile Workers

Intra company and Cloud


94/100

IM

Corp HQ

Corporate

Network

Presence

Call

Control

IVRIM

Office A

Presence

Call

Control

Voicemail

IM

Office B

Presence

Call

Control

PSTN

Internet

Intra-company and Cloud

Inter company Federation


95/100

UCSystem

Corp HQ

UCSystem

Office A

Corporate

Network

Company A

UCSystem

Corp HQ

UCSystem

Office A

Corporate

Network

Company B

Internet

Inter-company Federation


96/100

Resources

Some Good SIP Webpages


97/100

Some Good SIP Webpages

Tech-Invite (www.tech-invite.com) is a great source for general SIPinformation with many beautiful color illustrations

SIP Forum has many links to more info (www.sipforum.org)

RFC 5411, The Hitchhikers Guide to SIP(tools.ietf.org/html/rfc5411)

SIP Tutorial at iptel.org, (www.iptel.org/tutorial)

IETF information can be found at www.ietf.org and www.softarmor.com

P2PSIP.org (www.p2psip.org) is the community site for P2PSIP

Speaking of Standards blog (blogs.voxeo.com/speakingofstandards)

Books on SIP


98/100

Books on SIP

Internet Communications Using SIP, HenrySinnreich and Alan B. Johnston, Wiley NetworkingCouncil Series, 2006 (second edition)

SIP: Understanding the Session Initiation

Protocol, Alan B. Johnston, Artech House, 2004

SIP Beyond VoIP, Henry Sinnreich, Alan B.Johnston and Robert J. Sparks, VON Publishing,2005

SIP Demystified, Gonzalo Camarillo, McGraw-HillTelecom Series, 2002

Acknowledge/Thanks


99/100

Acknowledge/Thanks

Thanks to those who have worked on SIPover the years, presented SIP tutorials, orwritten SIP books!


100/100

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