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Introduction to SIP Russ Clark September 8, 2008
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Page 1: Introduction to SIP - Georgia Institute of Technology · Introduction to SIP Russ Clark September 8, 2008. IP Telephony. IP Telephony Voice over IP - VoIP

Introduction to SIP

Russ ClarkSeptember 8, 2008

Page 2: Introduction to SIP - Georgia Institute of Technology · Introduction to SIP Russ Clark September 8, 2008. IP Telephony. IP Telephony Voice over IP - VoIP

IP Telephony

Page 3: Introduction to SIP - Georgia Institute of Technology · Introduction to SIP Russ Clark September 8, 2008. IP Telephony. IP Telephony Voice over IP - VoIP

IP Telephony

Voice over IP - VoIP

• We want to take the analog, connection-oriented voice service and provide it over a digital, best-effort, datagram service.

• Are we crazy or what?

Page 4: Introduction to SIP - Georgia Institute of Technology · Introduction to SIP Russ Clark September 8, 2008. IP Telephony. IP Telephony Voice over IP - VoIP

IP Telephony

Voice over IP - VoIP

• We want to take the analog, connection-oriented voice service and provide it over a digital, best-effort, datagram service.

• Are we crazy or what?

• Must deal with Analog to Digital Conversion

• Codecs: e.g. G.711 - 64Kbps, G.729A - 8Kbps, G.723.1 - 6.4Kbps

Page 5: Introduction to SIP - Georgia Institute of Technology · Introduction to SIP Russ Clark September 8, 2008. IP Telephony. IP Telephony Voice over IP - VoIP

IP Telephony

Voice over IP - VoIP

• We want to take the analog, connection-oriented voice service and provide it over a digital, best-effort, datagram service.

• Are we crazy or what?

• Must deal with Analog to Digital Conversion

• Codecs: e.g. G.711 - 64Kbps, G.729A - 8Kbps, G.723.1 - 6.4Kbps

• Must deal with Delay, Jitter, Packet Loss

• <150 ms is imperceptible, 150-400 ms is OK, >400 ms is way out

Page 6: Introduction to SIP - Georgia Institute of Technology · Introduction to SIP Russ Clark September 8, 2008. IP Telephony. IP Telephony Voice over IP - VoIP

IP Telephony

Voice over IP - VoIP

• We want to take the analog, connection-oriented voice service and provide it over a digital, best-effort, datagram service.

• Are we crazy or what?

• Must deal with Analog to Digital Conversion

• Codecs: e.g. G.711 - 64Kbps, G.729A - 8Kbps, G.723.1 - 6.4Kbps

• Must deal with Delay, Jitter, Packet Loss

• <150 ms is imperceptible, 150-400 ms is OK, >400 ms is way out

• Must create the illusion of a call session

• need a session protocol

Page 7: Introduction to SIP - Georgia Institute of Technology · Introduction to SIP Russ Clark September 8, 2008. IP Telephony. IP Telephony Voice over IP - VoIP

VoIP Protocols

Page 8: Introduction to SIP - Georgia Institute of Technology · Introduction to SIP Russ Clark September 8, 2008. IP Telephony. IP Telephony Voice over IP - VoIP

VoIP Protocols

• SIP - Session Initiation Protocol - RFC 3261

• Call Managemt, Call Setup and Control

Page 9: Introduction to SIP - Georgia Institute of Technology · Introduction to SIP Russ Clark September 8, 2008. IP Telephony. IP Telephony Voice over IP - VoIP

VoIP Protocols

• SIP - Session Initiation Protocol - RFC 3261

• Call Managemt, Call Setup and Control

• SDP - Session Description Protocol - RFC 2327

• Describe the parameters for the voice session, carried in SIP INVITE

Page 10: Introduction to SIP - Georgia Institute of Technology · Introduction to SIP Russ Clark September 8, 2008. IP Telephony. IP Telephony Voice over IP - VoIP

VoIP Protocols

• SIP - Session Initiation Protocol - RFC 3261

• Call Managemt, Call Setup and Control

• SDP - Session Description Protocol - RFC 2327

• Describe the parameters for the voice session, carried in SIP INVITE

• RTP - Real Time Transport Protocol - RFC 3550

• For the data exchange, a stream of encoded voice packets

Page 11: Introduction to SIP - Georgia Institute of Technology · Introduction to SIP Russ Clark September 8, 2008. IP Telephony. IP Telephony Voice over IP - VoIP

VoIP Protocols

• SIP - Session Initiation Protocol - RFC 3261

• Call Managemt, Call Setup and Control

• SDP - Session Description Protocol - RFC 2327

• Describe the parameters for the voice session, carried in SIP INVITE

• RTP - Real Time Transport Protocol - RFC 3550

• For the data exchange, a stream of encoded voice packets

• RSVP - ReSerVation Protocol - RFC 2205

• Establish Priority and Reservations INSIDE the network

Page 12: Introduction to SIP - Georgia Institute of Technology · Introduction to SIP Russ Clark September 8, 2008. IP Telephony. IP Telephony Voice over IP - VoIP

VoIP Protocols

UA

UA

SIPServer

•User Agents (UA) connected over an IP network

Page 13: Introduction to SIP - Georgia Institute of Technology · Introduction to SIP Russ Clark September 8, 2008. IP Telephony. IP Telephony Voice over IP - VoIP

VoIP Protocols

UA

UA

SIPServer

SIP

•A SIP server isn’t technically required.•But we always use it. •Why? To find you.

Page 14: Introduction to SIP - Georgia Institute of Technology · Introduction to SIP Russ Clark September 8, 2008. IP Telephony. IP Telephony Voice over IP - VoIP

VoIP Protocols

UA

UA

SIPServer

SIP SDP

•SDP is carried as payload in the SIP INVITE and response

Page 15: Introduction to SIP - Georgia Institute of Technology · Introduction to SIP Russ Clark September 8, 2008. IP Telephony. IP Telephony Voice over IP - VoIP

VoIP Protocols

UA

UA

SIPServer

SIP

RTP

SDP

•RTP may not follow the same path as SIP

Page 16: Introduction to SIP - Georgia Institute of Technology · Introduction to SIP Russ Clark September 8, 2008. IP Telephony. IP Telephony Voice over IP - VoIP

VoIP Protocols

UA

UA

SIPServer

SIPRTP

RSVP

SDP

•RSVP is internal to the network•Current IMS implementations are limited here


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