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By Stephen Tomko
H.323 vs. SIP
Internal PBX CallExtension number is dialedPBX receives extensionRoutes extensionRoutes call to the phoneCall begins
Internal VoIP CallExtension number is dialedServer picks up extension
Tells phone how to reach the other phone via IP
VoIP phone establishes a connection to the other phoneVia TCP/IP protocol stackUses network mediums
Both confirm connection and call begins
Difference?Extension is still dialedCall is still placedStill routedCall quality is equal
ConvergenceData, voice, and video are all combined
under one mediumMany reliable mechanisms to provide
reliability, security, and manageability already existTCP, RTP, QoS paramsSSL, SSH, S-HTTP
How we handle callsVoIP, Video, and data must have sessionsSessions need protocols
Think SS7 for landlineTwo largest protocols
SIPH.323
SIP BackgroundCreated by the Internet Engineering Task
Force (IETF) as a method to control “sessions” between one or many pointsFebruary 1996
Designed around the HTTP protocolSession Codes
E.g. 400 – user error, 500 – server error, etc…
H.323 BackgroundCreated by ITU-T as a method to control
voice, audio, and dataNovember 1996
A Suite of protocolsH.225H.235H.245
Derived from Q.931 standardThink PSTN and ISDN signaling
SIP Components User Agent Client
Creates and sends requestsA SIP compatible phone is a User Agent doing the work of
a User Agent Client Registar/Location Server
Registers User AgentsStores locationsAddress resolution
(SIP:user2209@statefarmIT) User Agent Server
Accepts, Forwards and routes callsProxy Server
Routes calls Redirect Server
Multifunctioned Discussed Later
H.323 ComponentsTerminal
Standard VoIP phone or any device that starts or terminates H.323 sessions
GatewayTranslates data from one incompatible network to
anotherThink PSTN to Ethernet
GatekeeperControls calls and sessionsAddress resolutionZone controlBandwidth control
Multipoint Control UnitBridges many sessions into one
pulver.com
Proxy server
nortel.com
1. INVITE sip:[email protected] SIP/2.0 From: sip:[email protected]
Location Server
jeff
.pulv
er
pulv
er@
von1
2. INVITE sip:pulver@von1 SIP/2.0 From: sip:[email protected]. SIP/2.0 200 ok From: sip:pulver@von1
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]
1. INVITE sip:[email protected] From: sip:[email protected]
2. SIP/2.0 320 Moved temporarily Contact: sip:[email protected]
nortel.com
pulver.com
Redirect Server
Location Server
Jeff
.pulv
er
pulv
er@
von1
4. INVITE sip:[email protected] From: [email protected]
3. ACK sip:[email protected] From: sip:[email protected]
5. SIP/2.0 200 OK To: [email protected]
6. ACK sip:[email protected] From: sip:[email protected]
Pulver@von1
H.323 Process ExtensiveH.225 – Call signaling
Provides call signaling, alerting, and connected statuses for the call in question
RAS SignalingCommunication between terminals and gatekeepersCommunication between gatekeepers
H.245Capability negotiation
Defines what codecs to be used for audio/video/dataH.264, G.729, T.140
Master/Slave determinatorSettles all disputes between two devices during
negotiationH.235
Encryption using SSL
H.323 vs. SIPhttp://www.microtronix.ca/sip_vs_h323.htm
Business ValueConverging data/voice/video under one
medium - networkControl how your business communicates
Multipoint conferencingUnified communication allows all types of
communication under one protocolAll in one package
Why technology is importantOne word: Convergence umbrellaGeographic locations are reducedH.323
All-in-one suite, managers many facetsSIP
Simplicity, small, modularity
Major Commercial productsNortel’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 SIPhttp://www.avaya.com/usa/product/avaya-aura#
%20Avaya%20Aura%E2%84%A2%20Communication%20Manager
Cisco UT Communications Manager (CallManager) for H.323/SIPhttp://www.cisco.com/en/US/products/sw/
voicesw/ps556/