SIP and VOCAL

1
SIP and VOCAL

• David Bryan Cheung Tam
• Jasomi Networks, Inc. Cisco Systems, Inc.
• dbryan_at_jasomi.com cktam_at_cisco.com

2
VOCAL System

• The Vovida Open Communications Applications
  Library (VOCAL)
• Carrier grade VoIP system
• 80 person-years and millions of of
  development
• 1.8 Million Busy Hour Call Attempts
• Features and functions that you can use today
• FreeBSD-like license

3
VOCAL System

• Elevator Statement

VOCAL is the software that enables a core
network to support a Voice over IP system.
4
Traditional Telcos

• Proprietary Favoring the vendor over the
  customer
• Vendor-centric, limited interoperability
• Slow innovation, long development cycles
• Expensive and exclusive

5
Voice over IP

• Open standards, rapid development and greater
  choices for customers
• Packet routing more efficient use of network
  resources
• Network management less expensive to operate one
  network
• Feature development web-based applications are
  being developed and deployed quickly

6
Open Source VoIP

• Cheap and accessible. Developers can create the
  following
• New features that dont exist on the PSTN
• Enhanced functions and solutions that can
  accelerate the general adoption of VoIP in
  businesses and residences
• Processes and components that encourage others to
  express new ideas

7
Why Does Cisco Support Open Source VoIP?

• Its good for business
• Puts packets on the network Helps sell gateways
  and routers
• Distributive technology
• Helps win over new customers who want to control
  their destiny and their application source code

8
Voice over IP Protocols

• Pictorial Overview

9
SIP, H.323 and MGCP
Call Control and Signaling
Signaling and Gateway Control
Media
Audio/ Video
H.323
H.225
RTCP
RTP
MGCP
Q.931
RAS
SIP
H.245
RTSP
TCP
UDP
IP
H.323 Version 1 and 2 supports H.245 over TCP,
Q.931 over TCP and RAS over UDP. H.323 Version 3
and 4 supports H.245 over UDP/TCP and Q.931 over
UDP/TCP and RAS over UDP. SIP supports TCP and
UDP.
10
Session Initiation Protocol

11
What is SIP?

Session Initiation Protocol - An application
layer signaling protocol that defines initiation,
modification and termination of interactive,
multimedia communication sessions between users.

IETF RFC 2543 Session Initiation Protocol
12
SIP is Simply...

• An ASCII protocol, based on HTTP and SMTP, that
  enables
• Initializing, establishing and tearing down call
  sessions
• Building flexible distributed systems
• Interoperating with other web-based applications

13
Where Its _at_

• The SIP address is identified by a SIP URL, in
  the format user_at_host.
• Examples of SIP URLs
• siphostname_at_vovida.org
• siphostname_at_192.168.10.1
• sip14085551212_at_vovida.org

14
From 2 Devices...
SIP Components
15
...to a Distributed Network...
PSTN
User Agent
Gateway
16
...to a Complete Phone System
Phone
Gateway
Gateway
Marshal Server (Gateway)
CDR Server
Feature Server
Redirect Server
Provisioning Server
Policy Server
Marshal Server
Internet
Clearing House
Marshal Server
Marshal Server
H.323/SIP Translator
H.323 Terminal
MGCP/SIP Translator
Marshal Server
MGCP Device
17
VOCALVovida Open CommunicationsApplications
Library

18
VOCAL Say What?

• A library of communications applications that
  provides
• Open source software modules Primarily
  SIP-based, with some rough translators to H.323
  and MGCP endpoints
• A tool kit Building blocks that allow
  development of new VoIP features, applications
  and services

19
A Simplified Call Flow
The Audio Path is established.
8. User As phone responds with an ACK
(acknowledgement message)
7. User B, by picking up the phone, sends a 200
(OK) message.
6. User Bs phone rings and sends a 180 (Ring)
message back to User As phone.
5. Marshal Server B sends an INVITE message to
User B.
4. Marshal Server A forwards an INVITE message to
Marshal Server B.
3. The Redirect Server responds with routing
information and instructs the Marshal Server A
that User B can be contacted via the Marshal
Server B.
2. Marshal Server A forwards the INVITE message
to the Redirect Server
1. User A calls User B. User As Phone sends an
INVITE message to Marshal Server A.
20
System Diagram
Phone
Gateway
Gateway
Marshal Server (Gateway)
CDR Server
Feature Server
Redirect Server
Provisioning Server
Policy Server
Marshal Server
Internet
Clearing House
Marshal Server
Marshal Server
H.323/SIP Translator
H.323 Terminal
MGCP/SIP Translator
Marshal Server
MGCP Device
21
(No Transcript)
22
Redundancy
Phone
Gateway
Gateway
Marshal Server (Gateway)
CDR Servers
Feature Servers
Redirect Servers
Provisioning Server
Policy Servers
Marshal Server
Internet
Clearing House
Marshal Server
Marshal Server
H.323/SIP Translator
H.323 Terminal
MGCP/SIP Translator
Marshal Server
MGCP Device
23
Block Diagram
24
Example 1
Build your own soft phone
CSPS or partners
SIP Proxy
Soft phone
API
SIP
VOCAL SIP UA
SIP
Gateway
API
VOCAL SIP Stack
Phone
25
Example 2
Build your own CPL features
CPL scripts
API
CSPS or partners
VOCAL Feature Server
SIP Proxy
API
Gateway
SIP
VOCAL SIP Stack
SIP
SIP
Phone
26
Example 3
Build your own Routing Engine
Routing Logic
API
VOCAL Redirect Server
API
VOCAL SIP Stack
Invite
Gateway
Redirect
Phone
27
Future Features

• How SIP will be used
• Phone Calls (IPlt-gtIP and IPlt-gtPSTN)
• Instant Messaging
• Presence
• Games
• Whats missing
• IVR
• Voice Recognition
• Auto attendant

28
WIFY?

• Whats in it for you
• Build applications
• Add voice services to your ISP
• Build and manage your own PBX

29
Whos Working on it?

• Companies
• Dialpad
• Pagoo
• Cathay
• Netspeak
• Jasomi
• Lurkers

• Individuals
• Telecom engineers
• Students
• Hobbyists
• System Integrators
• IT

30
How Big is the Community?

• Recent Software Downloads

31
Vovida.org

• Mailing List Vocal_at_vovida.org
• Product Manager dhawk_at_cisco.com
• A few of our many contributors

32
www.vovida.org

• David Bryan Cheung Tam
• Jasomi Networks Cisco Systems, Inc
• dbryan_at_jasomi.com cktam_at_cisco.com

33
Additional Material

34
Vocal Components

35
6 Degrees of Separation

• Basic SIP calling process
• Registering, initiating and locating users.
• Determining acceptable media negotiating
  session descriptions between users.
• Determining the willingness of the called party
  to communicate calls are accepted or rejected
  through response messages.
• Setting up accepted calls.
• Modifying or handling calls through features
  for example, call transfer.
• Terminating the call.

36
VOCAL Components

• Redirect Server SIP-based location, registration
  and routing services.
• Provisioning Server Stores all subscriber and
  server data.
• Feature Server Uses CPL to provide basic
  telephony features including

37
VOCAL Components (2)

• Marshal Server Performs authentication and
  collects call detail records.
• CDR Server Collects billing records from
  Marshals, buffers them, and talks RADIUS to 3rd
  part billing systems.
• Policy Server Uses COPS to enable RSVP on
  routers. Uses OSP to communicate settlement info
  with VoIP clearing houses.

38
VOCAL Components (3)

• H.323 and MGCP Translators Line side translation
  from H.323 or MGCP endpoints into a SIP network.
• SIP Stack Linux- and Solaris- based
  implementation of the SIP protocol.
• SIP User Agent Client-side agent to terminate
  voice calls.
• Other Protocol Stacks MGCP, RTP, COPS, RADIUS
  and TRIP

39
Redirect Server

• Provides SIP redirect, location, and registration
  services/functions.
• Stores contact and feature subscription data for
  all registered subscribers
• Stores and provides dial plan and routing
  information
• Feature and Marshal Servers forwards INVITE
  messages to the Redirect Server to obtain routing
  information to route a call

40
Provisioning Server

• Stores data on each user or server within the
  VOCAL system
• Accessible from a Java-based GUI via an Internet
  browser this allows you to configure the VOCAL
  system and to administer users
• Allows you to add and provision new modules for
  redundancy and scalability
• All data is stored in XML format in a Linux
  directory structure
• Uses subscribe notify method to distribute
  provisioning information throughout the VOCAL
  system
• Subscribers can also access their feature
  information from an Internet browser this
  allows subscribers to enable or disable certain
  features that they have subscribed to. (For
  example, call forwarding).

41
Marshal Server - Authentication

• The Marshal Server currently supports
  authentication of the user by
• Access control verification of IP address
  against an access list
• Digest authentication verification of username
  and password
• The Marshal Server obtains user information from
  the Provisioning Server and/or the Redirect
  Server to authenticate the user.

The Provisioning Server checks to see if
userA_at_hostname.com is an authorized user.
Provisioning Server
2.REGISTER
3. 200 (OK)
Once userA_at_hostname.com has been registered, the
Redirect Server sends a 200 message (OK).
1. REGISTER
4. 200 (OK)
42
Marshal Server - Billing

• Billing
• Each Marshal server collects call detail record
  information (such as the start and stop of the
  call) and forwards CDR data to 3rd party billing
  systems using the Remote Authentication Dial In
  User Server (RADIUS) accounting protocol.

43
Feature Server
Supports new feature scripting using C and Call
Processing Language (CPL), an XML based language
for IP telephony features

• VOCAL-supported core network features
• Call Blocking
• Caller ID Blocking
• Call Forward All
• Call Forward No Answer
• Call Forward Busy
• Call Return
• Call Screen

• VOCAL-supported set- based features
• Transfer
• Caller ID
• Call Waiting
• 3 Way Calling

• VOCAL-supported applications
• Unified Messaging
• Conferencing

44
Features Scripting with CPL

• Call Processing Language (CPL) is
• An XML-based scripting language for describing
  Internet telephony services and creating end-user
  service features
• A lightweight language that is not a complete
  programming language it has no variables, loops
  or ability to run external programs
• A facility to enable decision making based on
  call properties such as time of day, calling
  party, called party and priority and action
  application such as call forwarding, call
  blocking, redirecting calls and sending e-mail
• An IETF draft

45
Call Detail Record (CDR) Server

• The Call Detail Record (CDR) Server
• Collects billing data from Marshal Servers
• Stores CDR for each call
• Forwards billing data to 3rd party billing system
  using Remote Authentication Dial In User Server
  (RADIUS) accounting protocol over UDP.
• A third-party billing system can generate
  invoices from the billing data.

46
Policy Server

• Functions
• To administer admission request for bandwidth and
  Quality of Service (QoS)
• Acts as a Policy Decision Point (PDP)
• Provides Internetworking Marshal Server with
  information on enforcing QoS
• Provides Policy Enforcement Points (PEP)
  routers or gateways with information or
  requests to reserve bandwidth for the duration of
  a specific call
• Communicates using
• Common Open Policy Service (COPS) protocol -
  used to communicate authorization
  request/responses between a Policy Server and
  Internetwork Marshal Servers or PEPs
• Open Settlement Protocol (OSP) used to exchange
  authentication, authorization, pricing and
  accounting information with clearinghouse servers
  when there are multiple VoIP network or service
  providers.

47
Scalability and Redundancy

• Scalability
• Modular approach - Redirect, Marshal and Feature
  Servers can be added in parallel
• These all use a round robin load balancing
  between all the active servers within any group
• Redundancy
• Active servers exchange multicast heartbeat info
  every 250 ms
• If a server drops, no established calls will be
  lost. Calls in an intermediate state may be lost
• If a server loses heartbeat with another server
  in the system, it will simply route messages to
  another server within the same group
• Redirect and Provisioning Servers maintain data
  and will synchronize before coming on-line

48
Session Initiation Protocol

49
What is SIP?

Session Initiation Protocol - An application
layer signaling protocol that defines initiation,
modification and termination of interactive,
multimedia communication sessions between users.

IETF RFC 2543 Session Initiation Protocol
50
SIP Framework
Instant Messaging

• Session initiation.
• Multiple users.
• Interactive multimedia applications.

Personal Mobility
Distance Learning
Video Conferencing
Email
Voice Calls
MPEG, MP3, Audio, HTML,XML
51
SIP Distributed Architecture
SIP Components
PSTN
User Agent
Gateway
52
SIP Messages Methods and Responses
SIP components communicate by exchanging SIP
messages

• SIP Requests
• INVITE Initiates a call by inviting user to
  participate in session.
• ACK - Confirms that the client has received a
  final response to an INVITE request.
• BYE - Indicates termination of the call.
• CANCEL - Cancels a pending request.
• REGISTER Registers the user agent.
• OPTIONS Used to query the capabilities of a
  server.
• INFO Used to carry out-of-bound information,
  such as DTMF digits.

• SIP Responses
• 1xx - Informational Messages.
• 2xx - Successful Responses.
• 3xx - Redirection Responses.
• 4xx - Request Failure Responses.
• 5xx - Server Failure Responses.
• 6xx - Global Failures Responses.

53
SIP Headers

• SIP borrows much of the syntax and semantics from
  HTTP.
• A SIP messages looks like an HTTP message
  message formatting, header and MIME support.
• An example SIP header
• --------------------------------------------------
  ---------------
• SIP Header
• --------------------------------------------------
  ---------------
• INVITE sip5120_at_192.168.36.180 SIP/2.0
• Via SIP/2.0/UDP 192.168.6.215060
• From sip5121_at_192.168.6.21
• To ltsip5120_at_192.168.36.180gt
• Call-ID c2943000-e0563-2a1ce-2e323931_at_192.168.6.2
  1
• CSeq 100 INVITE
• Expires 180
• User-Agent Cisco IP Phone/ Rev. 1/ SIP enabled
• Accept application/sdp
• Contact sip5121_at_192.168.6.215060
• Content-Type application/sdp

54
SIP Addressing

• The SIP address is identified by a SIP URL, in
  the format user_at_host.
• Examples of SIP URLs
• siphostname_at_vovida.org
• siphostname_at_192.168.10.1
• sip14083831088_at_vovida.org

55
Process for Establishing Communication

• Establishing communication using SIP usually
  occurs in six steps
• Registering, initiating and locating the user.
• Determine the media to use involves delivering
  a description of the session that the user is
  invited to.
• Determine the willingness of the called party to
  communicate the called party must send a
  response message to indicate willingness to
  communicate accept or reject.
• Call setup.
• Call modification or handling example, call
  transfer (optional).
• Call termination.

56
Registration

• Each time a user turns on the SIP user client
  (SIP IP Phone, PC, or other SIP device), the
  client registers with the proxy/registration
  server.
• Registration can also occur when the SIP user
  client needs to inform the proxy/registration
  server of its location.
• The registration information is periodically
  refreshed and each user client must re-register
  with the proxy/registration server.
• Typically the proxy/registration server will
  forward this information to be saved in the
  location/redirect server.

Proxy/ Registration Server
SIP Messages REGISTER Registers the address
listed in the To header field. 200 OK.
57
Simplified SIP Call Setup and Teardown
302 (Moved Temporarily)
ACK
INVITE
Call Setup
302 (Moved Temporarily)
ACK
MediaPath
Call Teardown
58
H.323

59
What is H.323?

Describes terminals and other entities that
provide multimedia communications services over
Packet Based Networks (PBN) which may not provide
a guaranteed Quality of Service. H.323 entities
may provide real-time audio, video and/or data
communications.

ITU-T Recommendation H.323 Version 4
60
H.323 Framework

• H.323 defines
• Call establishment and teardown.
• Audio visual or multimedia conferencing.

61
H.323 Components
Circuit Switched Networks
62
H.323 Terminals

• H.323 terminals are client endpoints that must
  support
• H.225 call control signaling.
• H.245 control channel signaling.
• RTP/RTCP protocols for media packets.
• Audio codecs.
• Video codecs support is optional.

63
H.323 Gateway

• A gateway provides translation
• For example, a gateway can provide translation
  between entities in a packet switched network
  (example, IP network) and circuit switched
  network (example, PSTN network).
• Gateways can also provide transmission formats
  translation, communication procedures
  translation, H.323 and non-H.323 endpoints
  translations or codec translation.

64
H.323 Gatekeepers

• Gatekeepers provide these functions
• Address translation.
• Admission control.
• Bandwidth control.
• Zone management.
• Call control signaling (optional).
• Call authorization (optional).
• Bandwidth management (optional).
• Call management (optional).
• Gatekeepers are optional but if present in a
  H.323 system, all H.323 endpoints must register
  with the gatekeeper and receive permission before
  making a call.

65
H.323 Multipoint Control Unit

• MCU provide support for conferences of three or
  more endpoints.
• An MCU consist of
• Multipoint Controller (MC) provides control
  functions.
• Multipoint Processor (MP) receives and
  processes audio, video and/or data streams.

66
H.323 is an Umbrella Specification
Media H.261 and H.263 Video codecs. G.711,
G.723, G.729 Audio codecs. RTP/RTCP Media.
H.323
Data/Fax T.120 Data conferencing. T.38 Fax.

• Call Control and Signaling
• H.245 - Capabilities advertisement, media channel
  establishment, and conference control.
• H.225
• Q.931 - call signaling and call setup.
• RAS - registration and other admission control
  with a gatekeeper.

67
Other ITU H. Recommendation that work with H.323
Protocol
Description
68
H.323 Components and Signaling
H.225/RAS messages over RAS channel
H.225/RAS messages over RAS channel
H.225/Q.931 (optional)
H.225/Q.931 (optional)
Gatekeeper
H.245 messages (optional)
H.245 messages (optional)
H.225/Q.931 messages over call signaling channel
PSTN
H.245 messages over call control channel
Terminal

• H.245 A protocol for capabilities
  advertisement, media channel establishment and
  conference control.
• H.225 - Call Control.
• Q.931 A protocol for call control and call
  setup.
• RAS Registration, admission and status protocol
  used for communicating between an H.323
  endpoint and a gatekeeper.

69
Process for Establishing Communication

• Establishing communication using H.323 is a
  five-step process
• Call setup.
• Initial communication and capabilities exchange.
• Audio/video communication establishment.
• Call services.
• Call termination.

70
Simplified H.323 Call Setup

• Both endpoints have previously registered with
  the gatekeeper.
• Terminal A initiate the call to the gatekeeper.
  (RAS messages are exchanged).
• The gatekeeper provides information for Terminal
  A to contact Terminal B.
• Terminal A sends a SETUP message to Terminal B.
• Terminal B responds with a Call Proceeding
  message and also contacts the gatekeeper for
  permission.
• Terminal B sends a Alerting and Connect message.
• Terminal B and A exchange H.245 messages to
  determine master slave, terminal capabilities,
  and open logical channels.
• The two terminals establish RTP media paths.

1. ARQ
2. ACF
3. SETUP
4. Call Proceeding
5. ARQ
6. ACF
7.Alerting
8.Connect
H.245 Messages
RTP Media Path
RAS messages
Call Signaling Messages
Note This diagram only illustrates a simple
point-to-point call setup where call signaling is
not routed to the gatekeeper. Refer to the H.323
recommendation for more call setup scenarios.
71
Versions of H.323
Version
Reference for key feature summary
Date
H.323 Version 3
http//www.packetizer.com/iptel/h323/whatsnew_v3.h
tml
September 1999
72
References

• For more information on H.323, refer to
• ITU-T
• http//www.itu.int/itudoc/itu-t/rec/index.html
• Packetizer
• http//www.packetizer.com/iptel/h323/
• Open H.323
• http//www.openH323.org

73
Comparing

• SIP and H.323

74
Comparing SIP and H.323 - Similarities

• Functionally, SIP and H.323 are similar. Both SIP
  and H.323 provide
• Call control, call setup and teardown.
• Basic call features such as call waiting, call
  hold, call transfer, call forwarding, call
  return, call identification, or call park.
• Capabilities exchange.

75
Comparing SIP and H.323 - Strengths

• H.323 Defines sophisticated multimedia
  conferencing. H.323 multimedia conferencing can
  support applications such as whiteboarding, data
  collaboration, or video conferencing.
• SIP Supports flexible and intuitive feature
  creation with SIP using SIP-CGI (SIP-Common
  Gateway Interface) and CPL (Call Processing
  Language).
• SIP Third party call control is currently only
  available in SIP. Work is in progress to add
  this functionality to H.323.

76
Table 1 - SIP and H.323
77
Table 2 - SIP and H.323
Information
H.323
SIP
78
Table 3 - SIP and H.323
79
Table 4 SIP and H.323
80
Reference

• This section cites a document that provides a
  comprehensive comparison on H.323 and SIP
• Dalgic, Ismail. Fang, Hanlin. Comparison of
  H.323 and SIP for IP Telephony Signaling in
  Proc. of Photonics East, (Boston, Massachusetts),
  SPIE, Sept. 1999.
• http//www.cs.columbia.edu/hgs/papers/others/
  Dalg9909_Comparison.pdf

81
MGCP

• Media Gateway Control Protocol

82
What is MGCP?

Media Gateway Control Protocol - A protocol for
controlling telephony gateways from external call
control elements called media gateway controllers
or call agents.

IETF RFC 2705 Media Gateway Control Protocol
83
Components

• Call agent or media gateway controller
• Provides call signaling, control and processing
  intelligence to the gateway.
• Sends and receives commands to/from the gateway.
• Gateway
• Provides translations between circuit switched
  networks and packet switched networks.
• Sends notification to the call agent about
  endpoint events.
• Execute commands from the call agents.

Call Agent or Media Gateway Controller (MGC)
Call Agent or Media Gateway Controller (MGC)
SIP H.323
MGCP
MGCP
84
Simplified Call Flow

• When Phone A goes off-hook Gateway A sends a
  signal to the call agent.
• Gateway A generates dial tone and collects the
  dialed digits.
• The digits are forwarded to the call agent.
• The call agent determines how to route the call.
• The call agent sends commands to Gateway B.
• Gateway B rings phone B.
• The call agent sends commands to both gateways to
  establish RTP/RTCP sessions.

MGCP
MGCP
RTP/RTCP
Gateway A
Gateway B
85
MGCP Commands

• Call Agent Commands
• EndpointConfiguration
• NotificationRequest
• CreateConnection
• ModifyConnection
• DeleteConnection
• AuditEndpoint
• AuditConnection

• Gateway Commands
• Notify
• DeleteConnection
• RestartInProgress

86
Characteristics of MGCP

• MGCP
• A master/slave protocol.
• Assumes limited intelligence at the edge
  (endpoints) and intelligence at the core (call
  agent).
• Used between call agents and media gateways.
• Differs from SIP and H.323 which are peer-to-peer
  protocols.
• Interoperates with SIP and H.323.

87
MGCP, SIP and H.323

• MGCP divides call setup/control and media
  establishment functions.
• MGCP does not replace SIP or H.323. SIP and H.323
  provide symmetrical or peer-to-peer call
  setup/control.
• MGCP interoperates with H.323 and SIP. For
  example,
• A call agent accepts SIP or H.323 call setup
  requests.
• The call agent uses MGCP to control the media
  gateway.
• The media gateway establishes media sessions with
  other H.323 or SIP endpoints.

88
Example Comparison

• MGCP
• A user picks up analog phone and dials a number.
• The gateway notifies call agent of the phone
  (endpoint) event.
• The Call agent determines capabilities, routing
  information, and issues a command to the gateways
  to establish RTP/RTCP session with other end.

• H.323
• A user picks up analog phone and dials a number.
• The gateway determines how to route the call.
• The two gateways exchange capabilities
  information.
• The terminating gateway rings the phone.
• The two gateways establish RTP/RTCP session with
  each other.

3
2
Call Agent/Media Gateway Controller
5.RTP/RTCP
4
1
1
RTP/RTCP
Gateway A
Gateway B
89
What is Megaco?

• A protocol that is evolving from MGCP and
  developed jointly by ITU and IETF
• Megaco - IETF.
• H.248 or H.GCP - ITU.
• For more information, refer to
• IETF - http//www.ietf.org/html.charters/megaco-ch
  arter.html
• Packetizer - http//www.packetizer.com/iptel/h248/

90
References

• For more information on MGCP, refer to
• IETF
• http//www.ietf.org/rfc/rfc2705.txt?number2705

91
VOCAL Resources

• Web site www.vovida.org
• Mailing lists vocal_at_vovida.org
• sip_at_vovida.org

92
General VoIP Reference

• Pulver IP Telephony News
• http//www.pulver.com
• Internet Telephony
• http//www.internettelephony.com
• An overview poster of the SIP, MGCP, and H323
  protocols.
• http//www.protocols.com/voip/posvoip.pdf

93
SIP References

• For more information about SIP, refer to
• IETF
• http//www.ietf.org/html.charters/sip-charter.html
  
• Henning Schulzrinne's SIP page
• http//www.cs.columbia.edu/hgs/sip/

94
General VoIP Reference

• Pulver IP Telephony News
• http//www.pulver.com
• Internet Telephony
• http//www.internettelephony.com
• An overview poster of the SIP, MGCP, and H323
  protocols.
• http//www.protocols.com/voip/posvoip.pdf