Transcript project

H.323 vs. SIP
By Stephen Tomko
Internal PBX Call
 Extension number is dialed
 PBX receives extension
 Routes extension
 Routes call to the phone
 Call begins
Internal VoIP Call
 Extension number is dialed
 Server picks up extension
 Tells phone how to reach the other phone via IP
 VoIP phone establishes a connection to the other phone
 Via TCP/IP protocol stack
 Uses network mediums
 Both confirm connection and call begins
Difference?
 Extension is still dialed
 Call is still placed
 Still routed
 Call quality is equal
Convergence
 Data, voice, and video are all combined under one medium
 Many reliable mechanisms to provide reliability, security, and
manageability already exist
 TCP, RTP, QoS params
 SSL, SSH, S-HTTP
How we handle calls
 VoIP, Video, and data must have sessions
 Sessions need protocols
 Think SS7 for landline
 Two largest protocols
 SIP
 H.323
SIP Background
 Created by the Internet Engineering Task Force (IETF) as a
method to control “sessions” between one or many points
 February 1996
 Designed around the HTTP protocol
 Session Codes
 E.g. 400 – user error, 500 – server error, etc…
H.323 Background
 Created by ITU-T as a method to control voice, audio, and
data
 November 1996
 A Suite of protocols
 H.225
 H.235
 H.245
 Derived from Q.931 standard
 Think PSTN and ISDN signaling
SIP Components
 User Agent Client
 Creates and sends requests
 A SIP compatible phone is a User Agent doing the work of a User Agent Client
 Registar/Location Server
 Registers User Agents
 Stores locations
 Address resolution
 (SIP:user2209@statefarmIT)
 User Agent Server
 Accepts, Forwards and routes calls
 Proxy Server
 Routes calls
 Redirect Server
 Multifunctioned
 Discussed Later
H.323 Components
 Terminal
 Standard VoIP phone or any device that starts or terminates H.323
sessions
 Gateway
 Translates data from one incompatible network to another
 Think PSTN to Ethernet
 Gatekeeper




Controls calls and sessions
Address resolution
Zone control
Bandwidth control
 Multipoint Control Unit
 Bridges many sessions into one
1. INVITE sip:[email protected] SIP/2.0
From: sip:[email protected]
2. INVITE sip:pulver@von1 SIP/2.0
From: sip:[email protected]
3. SIP/2.0 200 ok
From: sip:pulver@von1
pulver.com
nortel.com
[email protected]
jeff.pulver
Location Server
pulver@von1
4. SIP/2.0 100 OK
From: sip:[email protected]
5. ACK sip:[email protected] SIP/2.0
From: sip:[email protected]
6. ACK sip:pulver@von1 SIP/2.0
From: sip:[email protected]
Proxy server
pulver.com
nortel.com
Jeff.pulver
Location Server
[email protected]
Pulver@von1
Redirect Server
1. INVITE sip:[email protected]
From: sip:[email protected]
2. SIP/2.0 320 Moved temporarily
Contact: sip:[email protected]
4. INVITE sip:[email protected]
From: [email protected]
5. SIP/2.0 200 OK
To: [email protected]
3. ACK sip:[email protected]
From: sip:[email protected]
6. ACK sip:[email protected]
From: sip:[email protected]
H.323 Process Extensive
 H.225 – Call signaling
 Provides call signaling, alerting, and connected statuses for the call in
question
 RAS Signaling
 Communication between terminals and gatekeepers
 Communication between gatekeepers
 H.245
 Capability negotiation
 Defines what codecs to be used for audio/video/data
 H.264, G.729, T.140
 Master/Slave determinator
 Settles all disputes between two devices during negotiation
 H.235
 Encryption using SSL
H.323 vs. SIP
 http://www.microtronix.ca/sip_vs_h323.htm
Business Value
 Converging data/voice/video under one medium - network
 Control how your business communicates
 Multipoint conferencing
 Unified communication allows all types of communication
under one protocol
 All in one package
Why technology is important
 One word: Convergence umbrella
 Geographic locations are reduced
 H.323
 All-in-one suite, managers many facets
 SIP
 Simplicity, small, modularity
Major Commercial products
 Nortel’s Application Server 5300 for SIP
 http://products.nortel.com/go/product_content.jsp?segId=0
&catId=null&parId=0&prod_id=66621&locale=en-US
 Avaya Aura for SIP
 http://www.avaya.com/usa/product/avaya-
aura#%20Avaya%20Aura%E2%84%A2%20Communication%
20Manager
 Cisco UT Communications Manager (CallManager) for
H.323/SIP
 http://www.cisco.com/en/US/products/sw/voicesw/ps556/