Avaya Intro SIP
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Transcript of Avaya Intro SIP
© 2005 Avaya Inc. All rights reserved. Avaya – Proprietary & Confidential. For Internal Use Only.
Session Initiation ProtocolIntroduction
Ben JenkinsSr. Product Manager
Converged Communications Server
April 11th, 2005
2© 2005 Avaya Inc. All rights reserved. Avaya – Proprietary & Confidential. For Internal Use Only.
What is Session Initiation Protocol (SIP)?http://www.ietf.org/html.charters/sip-charter.html
“SIP is an IETF application layer-protocol that can establish, modify, and terminate multimedia sessions”
– RFC 3261
Media agnostic– Voice, video, instant messaging, etc.
Media negotiation– Offer-Answer model
Similar to HTTP– Request-Response model– Text message-based protocol
• Easy to debug
Reuses other IETF protocols– UDP, TCP, TLS, DHCP, DNS, SDP, RTP, MIME, etc.
3© 2005 Avaya Inc. All rights reserved. Avaya – Proprietary & Confidential. For Internal Use Only.
History of SIP
1995: Work Begins
Feb 1999: Published as RFC 2543
April 2000: SIP/SIMPLE selected by 3gpp
Adoption was initially very slow– H.323 vs. SIP debate
– Accelerated with the support of Cisco, Microsoft, Nokia, etc.• Summer 2001: MS announces SIP as core of Windows XP
Today– 3 major IETF SIP working groups– 40+ SIP RFCs– 100+ SIP-related Internet Drafts– SIP products from nearly every major telecom vendor
4© 2005 Avaya Inc. All rights reserved. Avaya – Proprietary & Confidential. For Internal Use Only.
What is SIMPLE?http://www.ietf.org/html.charters/sipping-charter.html
SIP for Instant Messaging and Presence Leveraging Extensions
– IETF working group
Introduces “Presence” into communications state– Builds on RFC 3265– Now a standard: RFC 3856– Selected as basis for 3gpp networks & applications
5© 2005 Avaya Inc. All rights reserved. Avaya – Proprietary & Confidential. For Internal Use Only.
What is SIPPING?http://www.ietf.org/html.charters/sipping-charter.html
Session Initiation Protocol Project INvestiGation– IETF working group
• Chartered to document the use of SIP for several applications related to telephony and multimedia
SIPPING-19 refers to SIP Services Examples draft– draft-ietf-sipping-service-examples-08– 19 example telephony features implemented in SIP– Purpose is to ensure that basic features interoperate
Other SIPPING items– SIP Basic Call Flow Examples (RFC 3665)– Message Waiting Indication (RFC 3842)
6© 2005 Avaya Inc. All rights reserved. Avaya – Proprietary & Confidential. For Internal Use Only.
Example SIP Phone Bootstrap Process
Power on
Acquire IP address from DHCP
Query DHCP for TFTP or otherwise discover SIP proxy server address
Register and authenticate with SIP proxy server
Begin making and receiving phone calls
7© 2005 Avaya Inc. All rights reserved. Avaya – Proprietary & Confidential. For Internal Use Only.
Number Portability NirvanaThe SIP Address of Record (AOR)
SIP provides a single user identity, the “public address”– e.g. sip:[email protected] or sip:[email protected]
User identity maps to any number of devices
Hoteling and User Mobility are native to SIP
Dual-ModeMobile Phone
PDA
Softphone
Traditional phone
SIP Phone
Instant Messaging
8© 2005 Avaya Inc. All rights reserved. Avaya – Proprietary & Confidential. For Internal Use Only.
Components of SIP
User Agent– User Agent Client
• Generates and sends SIP requests and receives responses– User Agent Server
• Receives SIP requests and generates SIP responses
Registrar– Provides mapping of logical SIP addresses to physical SIP addresses
Location Service– Used by SIP Proxy or Redirect server to obtain the mapping from logical SIP
addresses to physical SIP addressesProxy Server
– Forwards SIP requests downstream and responses upstreamRedirect Server
– Generates 3xx responses directing clients to contact an alternate set of URIsPresence Server
– Acts as a Presence Agent or proxy server for SUBSCRIBE requests
Avaya incorporates all these functions in the SIP Enablement Services
9© 2005 Avaya Inc. All rights reserved. Avaya – Proprietary & Confidential. For Internal Use Only.
SIP Messages
Requests (Methods)
REGISTER– Register contact information
INVITE, ACK, CANCEL– Setting up sessions
BYE– Terminating sessions
OPTIONS– Querying servers about their
capabilities
SUBSCRIBE, NOTIFY (RFC 3265)
– Event notification framework
MESSAGE (RFC 3428)– Instant messages
Responses
1xx: Provisional– request received, continuing to process
the request
2xx: Success– the action was successfully received,
understood, and accepted
3xx: Redirection– further action needs to be take in order to
complete the request
4xx: Client Error– the request contains bad syntax or
cannot be fulfilled at this server
5xx: Server Error– the server failed to fulfill an apparently
valid request
6xx: Global Failure– the request cannot be fulfilled at any
server
10© 2005 Avaya Inc. All rights reserved. Avaya – Proprietary & Confidential. For Internal Use Only.
Major Difference between SIP and H.323
SIP can be used for more than telephony and video
SIP is text-based; H.323 is binary
Dial tone and ring-back tone is generated locally by the phone
Dial plan resides in the phone
Digits are not transmitted to the PBX until the phone completes digit collection based on dial plan
SIP clients can be represented by alpha-numeric id’s and passwords
SIP users can associate multiple devices with a single AoR
11© 2005 Avaya Inc. All rights reserved. Avaya – Proprietary & Confidential. For Internal Use Only.
Example Call FlowBob
INVITE sip:[email protected]
407 Proxy Authentication Required
180 Ringing
ACK sip:[email protected]
Alice
BYE sip:[email protected]
200 Ok
Bobhangs up
Proxy
ACK sip:[email protected]
INVITE sip:[email protected]
100 Trying
INVITE sip:[email protected]
180 Ringing
200 Ok
200 Ok
ACK sip:[email protected]
BYE sip:[email protected]
200 Ok
RTP
Bobanswers
12© 2005 Avaya Inc. All rights reserved. Avaya – Proprietary & Confidential. For Internal Use Only.
Dissecting SIP Signaling Messages
INVITE sip:[email protected] SIP/2.0Contact: <sip:[email protected]:5060;transport=tcp>Via: SIP/2.0/TCP 192.168.1.100:5060;branch=z9hG4bK74bf9Via: SIP/2.0/TLS b.example.com:5061;branch=z9hG4bK7lsjjfVia: SIP/2.0/TLS edge.example.com:5061;branch=z9hG4bK7a83jfRecord-Route: <sip:edge.example.com:5061;transport=tls>Record-Route: <sip:b.example.com:5061;transport=tls>Max-Forwards: 70From: Alice <sip:[email protected]>;tag=9fxced76slTo: Bob <sip:[email protected]>Call-ID: [email protected]: 1 INVITEContent-Type: application/sdpContent-Length: 151
v=0o=alice 2890844526 2890844526 IN IP4 192.168.1.100s=-c=IN IP4 192.0.2.101t=0 0m=audio 49172 RTP/AVP 0a=rtpmap:0 PCMU/8000
Request URI Physical Location of Originator
Hops traversed en route from originator to recipient Hops that must be
traversed for subsequent requests within a dialog (i.e, BYE, re-INVITE)
AoR of Originator (ANI) and Recipient (DNIS)
Dialog identifiers and message sequence number within a dialog
Payload (body) meta-data
SDP Body
13© 2005 Avaya Inc. All rights reserved. Avaya – Proprietary & Confidential. For Internal Use Only.
SIP Transactions and DialogsBob
INVITE sip:[email protected]
407 Proxy Authentication Required
180 Ringing
ACK sip:[email protected]
Alice
BYE sip:[email protected]
200 Ok
Proxy
ACK sip:[email protected]
INVITE sip:[email protected]
100 Trying
INVITE sip:[email protected]
180 Ringing
200 Ok
200 Ok
ACK sip:[email protected]
BYE sip:[email protected]
200 Ok
RTP
Client T
ransaction
Dialo
gC
lient Tra
nsactionS
erver Tra
nsaction
Both transaction and dialog
14© 2005 Avaya Inc. All rights reserved. Avaya – Proprietary & Confidential. For Internal Use Only.
Supported IETF Standards & Drafts
RFC 1889 RTP: Real-Time Transport Protocol RFC 2246 The TLS Protocol RFC 2327 SDP: Session Description Protocol RFC 2396 URI generic syntax RFC 2617 Digest AuthenticationRFC 2782 DNS SRVRFC 2833 RTP Payload for DTMF DigitsRFC 3261 SIP: Session Initiation Protocol RFC 3262 Reliability of Provisional Responses in SIP RFC 3263 Locating SIP Servers RFC 3264 An Offer/Answer Model with SDP RFC 3265 SIP-Specific Event NotificationRFC 3311 UPDATE method†
RFC 3325 Private Extensions to SIP for Asserted Identity within Trusted NetworksRFC 3420 Internet Media Type message/sipfrag RFC 3428 SIP Extension for Instant Messaging RFC 3515 REFER methodRFC 3551 RTP Profile for Audio and Video Conferences with Minimal Control
RFC 3578 Mapping of ISDN User Part (ISUP) Overlap Signaling to SIPRFC 3840 Indicating User Agent Capabilities in SIP†
RFC 3841 Caller Preferences for the SIP†
RFC 3842 A Message Summary and Message Waiting Indication Event Package for SIPRFC 3856 A Presence Event Package for the SIPRFC 3891 The SIP "Replaces" HeaderRFC 4028 Session Timers in SIPdraft-elwell-sipping-redirection-reason-00draft-ietf-impp-cpim-03draft-ietf-impp-cpim-pidf-07draft-ietf-simple-winfo-format-04draft-ietf-simple-winfo-package-04draft-ietf-simple-presencelist-package-00draft-ietf-sip-history-info-03draft-ietf-sip-join-03draft-ietf-sipping-cc-conferencing-04draft-ietf-sipping-cc-transfer-02draft-ietf-sipping-conference-package-01draft-ietf-sipping-dialog-package-04
† Partial Support