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This chapter contains the following information about the Cisco SIP IP phone:
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 end points.
Like other VoIP protocols, SIP is designed to address the functions of signaling, device control, and session management within a packet telephony network. Signaling allows call information to be carried across network boundaries. Device control allows an intelligent entity to control a device with limited intelligence. Session management provides the ability to control the attributes of an end-to-end call.
SIP provides the capabilities to:
The conferences can consist of two or more users and can be established using multicast or multiple unicast sessions.
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Note The term conference means an established session (or call) between two or more end points. In this document, the terms conference and call are used interchangeably. |
SIP is a peer-to-peer protocol. The peers in the sessions are called User Agents (UAs). These user agents can be grouped into two categories:
Often a SIP end point will act as both a UAC and a UAS, one for sending requests the other for receiving.
From an architecture standpoint, the physical components of a SIP network can also be grouped into two categories: clients and servers. Figure 1-1 illustrates the architecture of a SIP network.
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Note In addition, the SIP servers can interact with other application services, such as Lightweght Directory Access Protocol (LDAP), a database application, or an extensible markup language (XML) application. These application services provide back-end services such as directory, authentication, and billing services. |
SIP clients include:
SIP servers include:
Cisco SIP IP phones 7960s (hereafter referred to as Cisco SIP IP phones) are full-featured telephones that can be plugged directly into an IP network and 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.
The Cisco SIP IP phone model terminals can attach to the existing in place data network infrastructure, via 10BaseT/100BaseT interfaces on an Ethernet switch. When used with a voice-capable Ethernet switch (one that understands Type of Service [ToS] bits and can prioritize VoIP traffic), the phones eliminate the need for a traditional proprietary telephone set and key system/PBX.
The Cisco SIP IP phone complies with RFC 2543.
Figure 1-2 illustrates physical features of the Cisco SIP IP phone:
In addition to the physical features illustrated in Figure 1-2, the Cisco SIP IP phone also provides the following:
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Note For information on how to use the standard telephony features and URL dialing, refer to the Getting Started Cisco IP Phone 7960 and Quick Reference Cisco IP Phone 7960 documents that shipped with the phone. |
The Cisco SIP IP phone supports the following standard network protocols:
For the Cisco SIP IP phone to successfully operate as a SIP endpoint in your network, your network must meet the following requirements:
The Cisco SIP IP phone has connections for connecting to the data network, for providing power to the phone, and for connecting a headset to the phone. Figure 1-3 illustrates the connections on the Cisco SIP IP phone.
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. On both the network port and access port, use full-duplex mode to avoid collisions.
Network Port (10/100 SW)
Use the network port to connect the phone to the network. You must use a straight-through cable on this port. The phone can also obtain inline power from the Cisco Catalyst switch over this connection. See the "Connecting to Power" section for details.
Use the access port to connect a network device, such as a computer, to the phone. You must use a straight-through cable on this port.
The Cisco SIP IP phone can be powered by the following sources:
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Note Only the network port (labeled 10/100 SW) supports inline power from the Cisco Catalyst switches. |
For redundancy, you can use the Cisco AC adapter even if you are using inline power from the Cisco Catalyst switches. The Cisco SIP IP phone can share the power load being used from the inline power and external power source. If either the inline power or the external power goes down, the phone can switch entirely to the other power source.
To use this redundancy feature you must set the inline power mode to auto on the Cisco Catalyst switch. Next, connect the un-powered Cisco SIP IP phone to the network. After the phone powers up, connect the external power supply to the phone.
The Cisco SIP IP phone supports a four or six-wire headset jack. Specifically, the Cisco SIP IP phone supports the following Plantronics headset models:
The Volume and Mute controls will also adjust volume to the earpiece and mute the speech path of the headset. The headset activation key is located on the front of the Cisco SIP IP phone.
To function in the IP telephony network, the Cisco SIP IP phone must be connected to a networking device, such as a Catalyst switch, to obtain network connectivity.
The Cisco SIP IP phone has an internal Ethernet switch, which enables it to switch traffic coming from the phone, access port, and the network port.
If a computer is connected to the access port, packets traveling to and from the computer and to and from the phone share the same physical link to the switch and the same port on the switch.
This configuration has these implications for the VLAN configuration on the network:
You can resolve these issues by isolating the voice traffic onto a separate VLAN on each of the ports connected to a phone. The switch port configured for connecting a phone would have separate VLANs configured for carrying:
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.
For more information, refer to the documentation included with the Cisco Catalyst switch.
Posted: Thu Aug 3 12:33:28 PDT 2000
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