Session Initiation Protocol
Session Initiation Protocol (SIP)
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Standards-based
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VoIP diagnostics (setup / teardown)
- Happens through SIP, might be Asterisk, might be UCM
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Carried by UDP 5060, also TCP or Stream Control Transmission Protocol (SCTP)
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Insecure by default
- You can see who is calling who
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Leverages Transport Layer Security (TLS) for privacy
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Establishes Realtime Transport Protocol (RTP) and Realtime Transport Control Protocol (RTCP) sessions.
Typical Phone Operation
- Phone boots via PoE connection
- Some switches can do this others can't
- DHCP offer will identify file server for phones
- DHCP offers IP, Subnet Mask, Gateway, DNS, TFTP
- TFTP is where we get the identity file for the phone which contains the configuration
- DHCP offers IP, Subnet Mask, Gateway, DNS, TFTP
- Phone OS & configuration are downloaded from file server
- Phone is ready
- SIP is used to establish an incoming or outgoing call
- Capabilities are negotiated and unidirectional audio channels are established between two parties using RTCP.
- They are gonna be monitored and manage
- If we see quality issues we can back off and lower the quality of the audio to see if it works better now.
- All happening on the background