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Cisco 7960g Administrator's Manual

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Product Overview
This chapter contains the following information about the Cisco SIP IP phone:
What Is Session Initiation Protocol?, page 1-1
•
•
What Is the Cisco SIP IP Phone?, page 1-3
Prerequisites, page 1-10
•
•
Cisco SIP IP Phone Connections, page 1-10
The Cisco SIP IP Phone with a Catalyst Switch, page 1-13
•
What Is Session Initiation Protocol?
Session Initiation Protocol (SIP) is the Internet Engineering Task Force's (IETF's) standard for
multimedia conferencing over IP. SIP is an ASCII-based, application-layer control protocol (defined in
RFC 2543) that can be used to establish, maintain, and terminate calls between two or more endpoints.
Like other VoIP protocols, SIP is designed to address the functions of signaling and session management
within a packet telephony network. Signaling allows call information to be carried across network
boundaries. Session management provides the ability to control the attributes of an end-to-end call.
SIP provides the capabilities to:
Determine the location of the target endpoint—SIP supports address resolution, name mapping, and
•
call redirection.
Determine the media capabilities of the target endpoint—Via Session Description Protocol (SDP),
•
SIP determines the "lowest level" of common services between the endpoints. Conferences are
established using only the media capabilities that can be supported by all endpoints.
Determine the availability of the target endpoint—If a call cannot be completed because the target
•
endpoint is unavailable, SIP determines whether the called party is already on the phone or did not
answer in the allotted number of rings. It then returns a message indicating why the target endpoint
was unavailable.
Establish a session between the originating and target endpoint—If the call can be completed, SIP
•
establishes a session between the endpoints. SIP also supports mid-call changes, such as the addition
of another endpoint to the conference or the changing of a media characteristic or codec.
•
Handle the transfer and termination of calls—SIP supports the transfer of calls from one endpoint
to another. During a call transfer, SIP simply establishes a session between the transferee and a new
endpoint (specified by the transferring party) and terminates the session between the transferee and
the transferring party. At the end of a call, SIP terminates the sessions between all parties.
OL-2206-01
C H A P T E R
Cisco SIP IP Phone Administrator Guide Version 3.0
1
1-1

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Summary of Contents for Cisco 7960g

  • Page 1 C H A P T E R Product Overview This chapter contains the following information about the Cisco SIP IP phone: What Is Session Initiation Protocol?, page 1-1 • • What Is the Cisco SIP IP Phone?, page 1-3 Prerequisites, page 1-10 •...

  • Page 2: Components Of Sip

    These application services provide back-end services such as directory, authentication, and billing services. Figure 1-1 SIP Architecture SIP proxy and redirect servers SIP user agents (UAs) SIP gateway PSTN Legacy PBX Cisco SIP IP Phone Administrator Guide Version 3.0 OL-2206-01...

  • Page 3: Sip Clients

    What Is the Cisco SIP IP Phone? Cisco SIP IP phones are full-featured telephones that can be plugged directly into an IP network and can be used very much like a standard private branch exchange (PBX) telephone. The Cisco SIP IP phone is an IP telephony instrument that can be used in VoIP networks.
  • Page 4 Cisco SIP IP phone: Figure 1-2 Cisco SIP IP Phone Physical Features LCD screen—Desktop, which displays information about your Cisco SIP IP phone, such as the • time, date, your phone number, caller ID, line and call status and the soft key tabs.

  • Page 5: Supported Features

    “Managing Cisco SIP IP Phones” section on page 3-1 for more information on configuration parameters. Ping support—Allows the user to use ping to see if a Cisco SIP IP phone is operational and how long • the response time is from the phone.
  • Page 6 • Multiple directory numbers—Allows the Cisco SIP IP phone to have up to six directory numbers or lines. Call waiting (enabled)—Plays an audible tone to indicate that an incoming call is waiting. The user •...
  • Page 7 • Call hold—Allows the Cisco SIP IP phone user (user A) to place a call (from user B) on hold. When user A places user B on hold, the two-way RTP voice path between user A and user B is temporarily disconnected, but the call session is still connected.
  • Page 8 The Domain Name Server RR (DNS SRV) is used to locate servers for a given service. SIP on Cisco’s SIP IP phones uses a DNS SRV query to determine the IP address of the SIP proxy or redirect server. The query string generated is in compliance with RFC 2782, and prepends the protocol label with an underscore _, as in “_protocol._transport.”...

  • Page 9: Supported Protocols

    DHCP allows you to move network devices from one subnet to another without administrative attention. If using DHCP, you can connect Cisco SIP IP phones to the network and become operational without having to manually assign an IP address and additional network parameters.

  • Page 10: Cisco Sip Ip Phone Connections

    The Cisco SIP IP phone supports UDP as it is defined in RFC 768 for SIP signaling. Prerequisites For the Cisco SIP IP phone to successfully operate as a SIP endpoint in your network, your network must meet the following requirements: A working IP network is established.

  • Page 11: Connecting To The Network

    (10/100 PC) Connecting to the Network The Cisco SIP IP phone has two RJ-45 ports that each support 10/100 Mbps half- or full-duplex Ethernet connections to external devices—network port (labeled 10/100 SW) and access port (labeled 10/100 PC). You can use either Category 3 or 5 cabling for 10 Mpbs connections, but use Category 5 for 100 Mbps connections.

  • Page 12: Using A Headset

    Only the network port (labeled 10/100 SW) supports inline power from the Catalyst switches. For redundancy, you can use the Cisco AC adapter even if you are using inline power from the Catalyst switches. The Cisco SIP IP phone can share the power load being used from the inline power and external power source.

  • Page 13: The Cisco Sip Ip Phone With A Catalyst Switch

    Voice traffic to and from the Cisco SIP IP phone (auxiliary VLAN) • Data traffic to and from the PC connected to the switch through the access port of the Cisco SIP IP • phone (native VLAN) Isolating the phones on a separate, auxiliary VLAN increases the quality of the voice traffic and allows a large number of phones to be added to an existing network where there are not enough IP addresses.
  • Page 14 Chapter 1 Product Overview The Cisco SIP IP Phone with a Catalyst Switch Cisco SIP IP Phone Administrator Guide Version 3.0 1-14 OL-2206-01...

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