Training program :

BACKGROUND CONTEXT TECHNOLOGY

Traditional telephony networks

IP telephony, the first step of the convergence of information systems (video / unified messaging voice / data / / sharing documents)

The reference models of ToIP

• Peer to Peer Model
• master / slave model
• architectures and components of NGN solutions (MGC (Media Gateway Controller) , MG (Media Gateway) , Signalling Gateway , IP -PBX , ... )

Coding (voice and video)

• the voice encoding ( G.711, G.729 , ... )
• video coding
• different transport networks ( ADSL , PPP , FR , Ethernet, ... )

Transport mechanism / voice RTP RTCP

TECHNOLOGY

Brief History of SIP

• standardization bodies ( IETF , SIP Forum ...)
• the reference model applicable to SIP (Peer to Peer)

Presentation of a basic SIP call

• with knowledge of the destination IP address
• without knowledge of the IP address of the recipient ( Registration and called using a proxy by the caller)
• release the call

The architecture and the components of SIP

Components

• User Agent (UAC , UAS)
• Registrar
• Proxy server (Stateless , Stateful , Forking ...)
• redirect Server
• rental server
• Using the DNS
• Gateway ( SIP / PSTN, SIP/H.323 , SIP / MGCP , ... )

The protocol stack (UDP, TCP , IP, SDP , RTP , DHCP, DNS)

• the syntax of SIP messages
•  the header structure / body
•  queries
• responses / errors
• header fields of header

Addressing

• URI
• E164
• ENUM : correspondence SIP address / phone number using DNS

Dynamic SIP

• transactions
• call flows
• call AU busy
• interrupted before dropping appeal
• call from a SIP phone to the PSTN via a gateway
• secure messages ( timers, Cseq , PRACK optional)

SDP for negotiating media streams

Using re-INVITE or UPDATE for re- negotiation of flow

Routing SIP

• using the field 'through'
• using the DNS proxy
• Keystone signaling field 'contact'
• Use headers 'Record- Route', 'route'

Services can add value with SIP (not exhaustive)

• Message Waiting Indicator (Message Waiting Indication)
• Conferences (Pre -arrange or Ad- hoc)
• Call Forwarding
• Warning / Hold
• Call termination on occupation
• Call Transfer
• Find Me / One number follow me
• Click to dial

Fax transmission and DTMF SIP

• DTMF inband and out- of-band
• FAX via T.38 or G.711

SIP security

• SIP and NAT
• SIP Firewall
• HTTP digest authentication
• encryption of RTP (SRTP) with SDP
• the signaling encryption : TLS (sips URI , using AES)
• managing encryption keys with MIKEY

Instant messaging and presence

• the use of SUBSCRIBE and NOTIFY messages (Agent co-located)
• using PUBLISH (Distributed Agent)
• the message SIMPLE MESSAGE

Comparative table of SIP / H.323, H.248 ( MEGACO ) , MGCP , proprietary protocols

Limitations of the SIP protocol , possible changes

List of the main RFC applicable to SIP

 

Subscribe to the INTER of your choice :

Ilexia adapts this training session INTRA Enterprise.

Illustrations & Demonstrations :

Connecting a SIP subscriber

• trace analysis (http digest authentication, registration ...)

Call between two SIP subscribers

• trace analysis

Connecting an IP PBX / SIP softswitch with a trunk and call a subscriber to subscriber PBX Softswitch

• trace analysis

Sending DTMF SIP phone

• trace analysis

 

Participants :

• Engineers, technicians and technical reponsables responsible for the operational management of enterprise networks
• Prerequisites: Experience & telecom networks
• Equipment provided: Support course paper

 

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