SIP Fundamentals for VoIP Professionals

SIP Fundamentals
VoiceCon Orlando 2010
Dan York, Director of Conversations, Voxeo

March 23, 2010

With thanks to David Bryan of Cogent Force

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

©© Voxeo Corporation (www.v o x e o .com) and David Bryan (www.e t h e r n o t .org)
Voxeo Corporation

About Dan York

www.ietf.org

www.voxeo.com
www.sipforum.org

blogs.voxeo.com/speakingofstandards

www.disruptivetelephony.com www.7ducattacks.com

Voxeo Corporation

About Voxeo

  Founded in 1999
  World’s largest hosted VoiceXML and CCXML platform – Over
82,000 hosted ports globally; hundreds of premise deployments
  Over 100,000 developers using Voxeo platforms
  The Voxeo difference: Unlocked Communications, Customer
Obsession Teams, Communications Passion
  Services/software available for free: www.voxeo.com/free

Voxeo Corporation

Voxeo - Built for VoIP Since Day One

  100% SIP VOIP-based
communications application
platform
•  Built from the ground up; natively
SIP-based
•  Other XML IVR platforms were built
on a CT card-centric model with SIP
added afterwards as a “virtual” or
“emulated” CT card
  Works with the widest variety of
SIP providers, gateways,
softphones and adapters
  SIPMethod SIP Servlets Platform
•  IM, Presence, Session Border
Controller

Voxeo Corporation

Voxeo: Design Once − Deploy Anywhere

Voxeo Corporation

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, particular
path not specified

  Original telephone network was circuit switched
(think of the operator w/wires)
•  SS7, defined by the ITU (International
Telecommunications Union)

Voxeo Corporation

Circuit Switched Network

There is a dedicated, fixed line (usually
physically connected) between parties, and
reserved for their use

Voxeo Corporation

Packet Switched Network

There is no physical connection. The call is
chopped into packets, which are sent over
a network of devices, perhaps using
different paths

Voxeo Corporation

Public Switched Telephone Network (PSTN)

Voxeo Corporation

Today...
ITSP
ITSP
ITSP
ITSP
ITSP

ITSP

ITSP
ITSP
Internet
ITSP
ITSP ITSP
Telephony
Service
Provider ITSP

ITSP PSTN ITSP
ITSP

ITSP
ITSP
ITSP ITSP
ITSP

ITSP
ITSP
ITSP

ITSP ITSP ITSP ITSP

ITSP ITSP
ITSP ITSP ITSP ITSP ITSP
Voxeo Corporation

Traditional Telephony
1.1

PSTN
PBX
Gateways

Physical
Voicemail
Wiring

Voxeo Corporation

VoIP and Uniﬁed Communications
1.2

Mobile
Devices
IM Application
Internet Servers
Networks
Operating
Systems
PSTN
IP-PBX VoIP
Gateways

Web IP Social
Firewalls
Servers Network Networks

Physical Directory
Voicemail
Wiring Servers

Desktop
Email PCs Database
Servers CRM Servers
Systems

Voxeo Corporation

VoIP Protocols

  H.323
•  ITU defined VoIP protocol
•  ITU is very formal, government oriented
•  More “telephony” in nature than Internet

  SIP
•  IETF (Internet Engineering Task Force) defined
protocol for general multimedia session establishment
•  IETF is much less formal, individual oriented
•  More “Internet” in nature than telephony

Voxeo Corporation

H.323 vs. SIP

Feature H.323 SIP
Encoding Binary Text
Control Centralized server
Endpoints (using
proxy)
Intelligence Most in central server; Most in the
core endpoints; edge
Modularity Monolithic (all in one Can be multiple
server) servers
Defined by ITU IETF
Media RTP RTP

Voxeo Corporation

SIP
  Session Initiation Protocol

  Defined (mostly) in IETF RFCs 3261-3264
•  Many other drafts extend SIP and add new
features

  Text based, very similar to HTTP, mail protocols

  Establishes a media session between endpoints

  Allows mobility – locate users using a SIP URI (sip:dyork@voxeo.com)
•  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 over
TCP/IPv6)

Voxeo Corporation

SIP

  Offers Event management (Subscribe/Notify)
for presence etc.
  Extended by SIMPLE to support IM (Instant
Messaging)
  Designed to allow easy interoperation with
other Internet features such as WWW and
email

Voxeo Corporation

Why is SIP Important?
  It’s all about standards

  The promise of interoperable equipment
•  A Nortel phone talking to a Cisco proxy to
reach 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

Voxeo Corporation

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 a
system
•  Redirect Server – used to inform devices when they need
to contact different locations
•  Proxy Server – used to relay messages back and forth
within the system

  In practice, several logical functions may actually
reside in the same physical server/program

Voxeo Corporation

User Agents

  A User Agent (also called a UA, Phone, or
Endpoint) 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

Hard Phone Mobile Hard Phone Soft Client
Voxeo Corporation

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 all of this

Voxeo Corporation

Basic SIP Call Flow
3.1

Proxy

SIP SIP

Alice Bob
Media

Voxeo Corporation

SIP & Media Paths

SIP SIP
Proxy Proxy
A SIP B

SIP SIP

Alice Bob
Media (RTP, MSRP, etc.)

Voxeo Corporation

Proxy
Bob is over at
bob-isp.com.
I’ll proxy the
call over there!

“I need to call Bob!”
sip.bob-office.com

“Call here for
Bob from Alice”

“Call for you
from Alice!”

sip.bob-isp.com

Voxeo Corporation

Registration

“I’m Alice, and I’ll be using a When I get a
phone at 192.168.0.1 today!” call for Alice, I’ll
know to
contact her at
192.168.0.1!
192.168.0.1 Registrar

sip.cogentforce.com

Voxeo Corporation

Registration

SIP
Registrar

1
SIP

2
SIP SIP
Proxy Proxy
A SIP B

SIP SIP

Alice Bob

Voxeo Corporation

Redirection

“I need to
call Bob!”

sip.cogentforce.com

“Call here for “Call here
Bob from Alice” for Bob
from Alice” sip.bob-isp.com
“Try him at
bob-isp.com”

sip.bob-office.com “Call for you
from Alice!”

Voxeo Corporation

The Simple Picture

SIP SIP
Proxy Proxy
A SIP B

SIP SIP

Alice Bob

Voxeo Corporation

The Complex Reality

SIP SIP SIP SIP SIP
Proxy Proxy Proxy Proxy Proxy
A SIP B SIP C SIP D SIP N

SIP Internet (or WAN) SIP

Media Media
Alice Proxy Proxy Bob
Media A Media B Media

Voxeo Corporation

SIP Messages

  SIP messages fall into two types – requests and
responses
  Requests include a method telling what it is they
are 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

Voxeo Corporation

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, 403
Forbidden
•  5xx Server Failure – 503 Service Unavailable
•  6xx Global Failure – 600 Busy Everywhere

Voxeo Corporation

Registration Call Flow

SIP Server

REGISTER

401 Unauthorized

REGISTER

200 OK

Voxeo Corporation

Basic Call Flow
SIP Server

INVITE
INVITE
100 Trying
180 Ringing
180 Ringing
200 OK
200 OK
ACK
ACK
Bi-directional Media between endpoints
BYE
BYE

200 OK
200 OK
Voxeo Corporation

No-answer Call Flow
SIP Server

INVITE
INVITE
100 Trying
180 Ringing
180 Ringing

CANCEL
200 OK
CANCEL
200 OK
487 Request Terminated
487 Request Terminated
ACK
ACK

Voxeo Corporation

AORs and Contacts

  In SIP, a registrar maps between an AoR and a
contact

•  AoR: Address of Record
•  Permanent, something on your business card
•  Not tied to a specific device
•  sip:dyork@voxeo.com
•  Contact
•  Ephemeral, tied to a device
•  sip:dyork@pc141.voxeo.com
•  Being able to change AoR→Contact mapping allows
mobility

Voxeo Corporation

Registration, revisited

“I’m Alice (alice@phonecompany.com), and
I’ll be using a phone at 192.168.0.1 today!”

192.168.0.1
Registrar

AoR → Contact Map:
alice@phonecompany.com → alice@192.168.0.1

Voxeo Corporation

Registration, revisited

“Lunch time! Mobile Time!
(alice@phonecompany.com)
will be using 10.0.1.1 now!”

Registrar
10.0.1.1

AoR → Contact Map:
alice@phonecompany.com → alice@10.0.1.1

Voxeo Corporation

INVITE message

INVITE sip:bob@bigcompany.com SIP/2.0
Via: SIP/2.0/TCP aliceua.phonecompany.com:5060
From: Alice <sip:alice@phonecompany.com>;tag=ss95cbav
To: Bob <sip:bob@bigcompany.com>
Call-ID: 3848276298220188511@aliceua.phonecompany.com
CSeq: 1 INVITE
Max-Forwards: 10
Contact: <sip:alice@aliceua.phonecompany.com;transport=tcp>
…snip!...

(Alice is going to call Bob…)

Voxeo Corporation

INVITE message
SIP request, method is INVITE, target is an AoR (for Bob)
Via header tracks where it has been (more on this later)
From: Alice <sip:alice@phonecompany.com>; tag=ss95cbav
Message is from Alice, and this is the SIP URL for Alice. The tag is unique
to this call and selected by Alice’s UA
Alice is trying to contact Bob at his SIP URL (a tag will be selected by
Bob’s UA later
Call-ID is a unique ID to track this particular call (selected by Alice’s UA)
CSeq: 1 INVITE
Used to track which responses go with which messages. Responses to
this invite will also have a CSeq of 1 INVITE
Max-Forwards: 10
Used in routing (more on this later)
How to contact Alice directly. Notice this is a contact (has a host)
…snip…
Voxeo Corporation

Calls and Dialogs

  A call is an informal term for communication
between two devices

  A dialog is 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

Voxeo Corporation

Transaction
  A transaction defines a particular operation that is
carried 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 new
transaction they initiate – no specified starting point
•  But since some transactions are initiated by each side,
within a call it might look out of order

Voxeo Corporation

Dialogs and Transactions
SIP Server

INVITE
INVITE
100 Trying
Transaction 1

180 Ringing
180 Ringing
200 OK
Dialog/Call

200 OK
ACK
ACK
Bi-directional Media between endpoints
Transaction 2

BYE
BYE

200 OK
200 OK
Voxeo Corporation

Routing

  Each entity forwards message along path to
reach destination
•  Places next destination into Request-URI (top line)
•  Adds themselves as another Via
•  Max-Forwards is decremented to prevent loops and
limit the length of path
•  if it reaches 0, the message isn’t forwarded again

Voxeo Corporation

INVITE message

CSeq: 1 INVITE
Max-Forwards: 10
…snip!...

Let’s see what happens as this message
passes through the big company proxy on the
way to Bob…

Voxeo Corporation

INVITE message

INVITE sip:bob@bobua.bigcompany.com SIP/2.0
Via: SIP/2.0/TCP proxy.bigcompany.com:5060
CSeq: 1 INVITE
Record-Route: <sip:proxy.bigcompany.com;lr>
Max-Forwards: 9
…snip!...
Notice the new URI, added VIA, and
decremented Max-Forwards. This message is
forwarded to Bob’s UA.

Voxeo Corporation

Routing Responses

  When Bob’s UA responds, it “unrolls” the Via
headers to decide where to route the responses

  The response traces the path back, sending the
response to the Via address of the entity that
sent the message (responding!)

  That entity removes itself from the Via and
passes the message along, unrolling until it
reaches the sender

  Notice that the proxy needs no state here

Voxeo Corporation

INVITE message

INVITE sip:bob@bobua.bigcompany.com SIP/2.0
Via: SIP/2.0/TCP proxy.bigcompany.com:5060
CSeq: 1 INVITE
Record-Route: <sip:proxy.bigcompany.com;lr>
Max-Forwards: 9
…snip!...

But one more here. What is record route?

Voxeo Corporation

Record Route

  Once a UA has the address of the other UA
in the call, new messages can be sent
directly between devices if they wish

  Record route is a way for a proxy to request
“keep me in the loop”
•  Ensures that future transactions include this proxy,
rather than going directly to the other UA

Voxeo Corporation

Forking

  While somewhat complicated, many features
require “forking”
•  Send a message to more than one entity Need to
track (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

Voxeo Corporation

Parallel Forking

time a
SIP Server Desk Phone

Caller time a

Two devices are tried at the same Mobile
time, time a

Voxeo Corporation

Sequential Forking

time a
SIP Server Desk Phone

Caller time b

One device is tried at time a, then Voice Mail
another at time b

Voxeo Corporation

Parallel & Sequential Forking

Desk Phone
time a

SIP Server time a

Mobile Phone

Caller
time b

Several devices are tried in
parallel at time a, then forked to
Voice Mail
another at time b

Voxeo Corporation

SIP Trunking & PSTN Interconnection

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 user’s proxy directly,
and the call is all SIP
•  To call out to a regular phone (or to have calls come
in) you use a gateway. It connects to one or more
regular phone lines to let you receive and place calls
to old fashioned “Plain Old Telephone
Service” (POTS)
•  Can also gateway between SIP and other protocols
such as H.323, or to networks such as Skype

Voxeo Corporation

Gateway example

SIP Server

Gateway

SIP/IP network POTS network

Voxeo Corporation

SIP Trunking

  Instead of using a local gateway, use SIP to
connect 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 IP
address of the ITSP.

  Can be over public Internet or over WAN, dedicated
lines or VPN
Voxeo Corporation

Trunking before SIP

Internet

Carrier PSTN
PBX

Corp HQ

Voxeo Corporation

SIP Trunking

ITSP

Internet
PBX

PSTN

Corp HQ

Voxeo Corporation

SIP Trunking – Multiple ITSPs

ITSP 1

Internet
PBX
ITSP 2
PSTN

Corp HQ

ITSP 3

Voxeo Corporation

SIP Trunking - Failover

PBX
ITSP

Corp HQ
Internet PSTN

PBX

Oﬃce
A

Voxeo Corporation

SIP Trunking - Geography

ITSP
(Boston)

Internet
PBX ITSP
(Paris)
PSTN

Corp HQ

ITSP
(Tokyo)

Voxeo Corporation

SIP Trunking - SIPconnect

  Initiative of the
SIP Forum to increase
interoperabilty between on-premise systems
and SIP service providers

  Idea is that a “SIPconnect-compliant” IP-PBX
will more easily be able to connect to a
“SIPconnect-compliant” SIP service provider

  http://www.sipforum.org/sipconnect

Voxeo Corporation

The Many Protocols Used In SIP

The (Many) Protocols in a “SIP” Call

UA UA
SIP Server
SIP/SDP SIP/SDP

Media (RTP/RTCP)

Voxeo Corporation

SIP, SDP, and RTP

  SIP is only a part of the picture
  SIP is used to establish the multimedia
session
  SIP messages don’t describe the type of
media themselves
•  Embed another protocol called SDP

  Yet another protocol, RTP (and RTCP), is
used to actually stream the media between
the devices

Voxeo Corporation

Codecs (encoding)

  The media that flows using RTP is encoded
using a codec
  Codec is about how the audio has been
converted 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 are
described in SDP

Voxeo Corporation

SDP/RTP

  Both of these are older IETF protocols used by
SIP
•  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

Voxeo Corporation

INVITE message

From: Alice <sip:alice@phonecompany.com>; tag=ss95cbav
Call-ID: 3848276298220188511@alicaua.phonecompany.com
CSeq: 1 INVITE
Content-Type: application/sdp
Content-Length: 134

v=0
o=CogentForce-UA 10010 605 IN IP4 aliceua.phonecompany.com
s=SIP Call
c=IN IP4 192.0.1.1
t=0 0
m=audio 49172 RTP/AVP 0 8 16

Voxeo Corporation

INVITE message

<snip!>
Content-Type: application/sdp
Content-Length: 134
The content of the message is SDP – to describe the media of the
call. We have 134 bytes of SDP
v=0
v is the SDP version number -- 0
o=CogentForce-UA 10010 605 IN IP4 aliceua.phonecompany.com
o is a description of the UA and a session ID
s=SIP Call
s is an string to encode a name for this session
c=IN IP4 192.0.1.1
c tells the other side what address to send the media to
t=0 0
t indicates start and end times. 0 and 0 don’t specify times in advance
m=audio 49172 RTP/AVP 0 8 16
m describes the media. We are listening for audio of type
RTP/AVP on port 49172
Voxeo Corporation

Offer/Answer

  Different UAs may support different codecs
  Calling UA lists what it supports and is willing to
use in the INVITE message SDP
  Called UA responds with the subset of that it is
willing to use in the response SDP (200 OK)
  If nothing in common, can’t communicate and
call is rejected
  Mechanisms exist to change this mid call (for
example, switching from audio to video)

Voxeo Corporation

Instant Message and Presence

  SIP has a set of extensions called
SIMPLE
•  SIMPLE: SIP for Instant Messaging
(IM) and Presence Leveraging
Extensions
•  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 (and
persistent on server)
•  Idle, Offline, Available, Busy, Offline
till Monday…

Voxeo Corporation

Instant Message and Presence

  MESSAGE method used to send text
messages
  SUBSCRIBE to express an interest,
“subscribe” to someone’s status
  PUBLISH to change your status on the
presence server

  NOTIFY is send from server to those who
have subscribed

Voxeo Corporation

Extending SIP
  SIMPLE isn’t 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 “standard”
extension
•  SIPCORE, DISPATCH, SIMPLE, and others

  Many vendors add non-interoperable “extensions” to SIP
•  These may not work across different vendor platforms

Voxeo Corporation

NAT/FW Traversal

  SIP has traditionally had some trouble getting
through firewalls (FWs) and Network Address
Translators (NATs)

  Two major causes
•  NATs use “private” IP addresses, not routable from
the 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 to
work and must be opened separately

Voxeo Corporation

Firewalls and Media

A UA Contacts
another UA (or a
proxy), sending SIP
messages.
The process of
sending the
message (on most
NATs/FWs) creates
a mapping that
allows responses to
reach the sender
SIP

Voxeo Corporation

Firewalls and Media

Media (RTP) is on a
different port
(different opening in
the FW) than SIP
It is possible that at
the time the media
is sent, it is rejected
by the firewall

RTP

Voxeo Corporation

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 hosts
behind firewalls
•  Session Border Controllers (SBCs)
•  Sit near (or in) the firewall/NAT, rewrite the internals of the
packets, work with firewall to open ports

Voxeo Corporation

Session Border Controllers

  Implemented either as special firewall/ALG that
modify the packets (deep-packet inspection) or as a
B2BUA
•  B2BUA can be thought of as a special SIP-to-SIP gateway
or as two UAs “glued together”, where each side is a
separate call

  Can be used for
•  NAT Traversal
•  Protocol “correction”
•  Demarcation point
•  Codec translation

  Generally handles both media and signalling

Voxeo Corporation

Session Border Controller

SIP Server SIP Server

Session Border Controller
(SBC)

SIP/IP network SIP/IP network

Voxeo Corporation

Peer-to-Peer (P2P) SIP

  Alternate way to deliver SIP

  Basic idea – reduce or completely eliminate
the central servers (proxy/softswitch)

  Most or all functionality is moved to the UA
  Highly scalable, so good for very small
deployments (no infrastructure) and
internet-wide deployments

Voxeo Corporation

Client/Server Session

In a Client/Server session, two nodes
must use a central server to
communicate

Voxeo Corporation

P2P Session

In a Peer-to-Peer session, when two
nodes communicate, a few other nodes,
rather than a central server, help
complete the call

Voxeo Corporation

Home Teleworkers
7.1

PC

UC Home
System
Firewall Internet Firewall

IP
Corp HQ Phone
Home

Voxeo Corporation

Mobile Workers

Laptop
UC
client
WiFi
UC
System
Firewall Internet Café
Router

Corp HQ
Mobile
Data
Network Mobile
UC
client

Voxeo Corporation

Intra-company and Cloud

Corporate
Internet
Network

IVR Voicemail
IM IM IM

Presence Presence Presence

Call Call Call
Control Control Control

Corp HQ Oﬃce A Oﬃce B

PSTN

Voxeo Corporation

Inter-company Federation

Internet

Corporate Corporate
Network Network

UC UC UC UC
System System System System

Corp HQ Oﬃce A Corp HQ Oﬃce A

Company A Company B

Voxeo Corporation

Some Good SIP Webpages

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

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

  RFC 5411, “The Hitchhiker’s 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)

Voxeo Corporation

Books on SIP

  Internet Communications Using SIP, Henry
Sinnreich and Alan B. Johnston, Wiley Networking
Council 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-Hill
Telecom Series, 2002

Voxeo Corporation

Acknowledge/Thanks

  Thanks to those who have worked on SIP
over the years, presented SIP tutorials, or
written SIP books!

Voxeo Corporation

SIP Fundamentals for VoIP Professionals

SIP Fundamentals for VoIP Professionals

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