Session Initiation Protocol

Session Initiation Protocol (SIP)

• Standards-based

• VoIP diagnostics (setup / teardown)

  • Happens through SIP, might be Asterisk, might be UCM

• Carried by UDP 5060, also TCP or Stream Control Transmission Protocol (SCTP)

• Insecure by default

  • You can see who is calling who

• Leverages Transport Layer Security (TLS) for privacy

• 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
• 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