Wednesday, December 15, 2010

SIP Third-Party IP Phone Support in CUCM Best CCIE Training Institute in New delhi Gurgaon

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CUCM supports RFC 3261–compliant third-party SIP phones. Support for third-party SIP
phone features varies greatly from Cisco SIP IP Phone features. Third-party phones have
only RFC 3261 SIP Version 2 support, whereas Cisco SIP Phones have many Cisco SCCP
features that have been rewritten to work in a native SIP protocol stack.
Two different types of third-party SIP phones can be added to CUCM. Third-party SIP
phones may be added as basic or advanced phones. Third-party SIP basic phones support
one line appearance and consume three DLUs. Third-party SIP advanced phones support
up to eight lines and video but consume six DLUs. Basic and advanced third-party SIP
phones offer the same telephony features.
Third-party SIP phones register with CUCM but do not use a MAC address–based device
ID. CUCM uses SIP digest authentication to identify a registering SIP third-party SIP
phone.
158 Chapter 7: Endpoints
Both CUCM and the third-party SIP phone have to be configured with a username and
password for digest authentication to work properly. CUCM refers to this item as a digest
user in which a user is associated with the phone in both the phone and user configuration
pages. The third-party device must also be configured with a matching username.
SIP standards and drafts supported by CUCM include the following:
■ RFC 3262: PRACK
■ RFC 3264: SDP offer/answer
■ RFC 3311: UPDATE
■ RFC 3515: REFER
■ RFC 3842: MWI Package
■ RFC 3891: Replaces Header
■ RFC 3892: Referred-by Mechanism
■ draft-levy-sip-diversion-08.txt: Diversion Header
■ draft-ietf-sip-privacy-04.txt: Remote-Party-Id Header
The following audio and video standards are supported for third-party SIP phones:
■ Audio:
—Audio codecs: G.711 mu-law, GSM Full-rate, G.723.1, G.711 A-law,
G.722, G.728, G.729
—RFC 2833: DTMF (Telephony-event)
■ Video:
—Video codecs: H.261, H.263, H.263+, H.263++, H.264
Cisco is working with key third-party vendors who are part of the Cisco Technology
Development Partner Program. These partners are developing solutions that leverage the
NOTE For more information about the support of these standards, see the Cisco SIP IP
Administrator Guide, Version 8.0 - Compliance with RFC 3261:
http://www.cisco.com/en/US/products/sw/voicesw/ps2156/products_administration_
guide_chapter09186a00807f47e3.html
SIP Third-Party IP Phone Support in CUCM 159
new CUCM and CUCM Express SIP capabilities. These vendors include Linksys,
IPCelerate, Research In Motion, IP blue, and Grandstream.
Cisco is also participating in an independent third-party testing and interoperability verification
process being offered by tekVizion. This independent service has been established
to enable third-party vendors to test and verify the interoperability of their endpoints with
CUCM and CUCM Express.
Third-party SIP phones support only a few features compared to Cisco IP Phones using
SCCP or SIP. The features that are not supported include but are not limited to the following:
■ MAC address registration
■ Phone buttons template
■ Soft key templates
■ Telephony features and applications such as the following:
—IP phone services
—CUCM Assistant
—Cisco Unified Video Advantage
—Call Pickup
—Barge
—Presence
SIP Third-Party Authentication
Digest authentication allows CUCM to act as a server to challenge the identity of a SIP
device when it sends a request to CUCM. When digest authentication is enabled for a
phone, CUCM challenges all SIP phone requests except keepalive messages. CUCM does
not support responding to challenges from SIP phones.
CUCM can challenge SIP devices connecting through a SIP trunk and can respond to
challenges received on its SIP trunk interface.
CUCM digest authentication is used to determine the identity of a third-party SIP phone.
The phones cannot be authenticated via MAC address like SCCP devices because thirdparty
SIP phones do not register by MAC address. Digest authentication is the industry
standard.
160 Chapter 7: Endpoints
CUCM can ignore the keyed hash that is provided in a digest authentication response and
check only if the provided username exists and is bound to a third-party SIP phone. This is
the default behavior. Alternatively, CUCM can be configured to check that the key that was
used at the third-party SIP phone to generate the keyed hash matches the locally configured
key (called digest credentials) at the end-user configuration in CUCM.
Third-party SIP phones cannot be configured by using the CUCM TFTP server. Instead,
they need to be configured using the native phone configuration mechanism, which is
usually a web page or a TFTP file. The device and line configuration in the CUCM database
must be manually synchronized with the native phone configuration (for example, extension
1002 on the phone and 1002 in CUCM). In addition, if the directory number of a line is
changed, it must be changed in both CUCM Administration and in the native phone
configuration mechanism.
Third-party SIP phones include their directory number in the registration message. They do
not send a MAC address. Third-party SIP phones identify themselves with digest authentication.
The SIP REGISTER message includes a header with a username and the keyed hash, as
shown in the following example:
Authorization: Digest
username=“3rdpsip”,realm=“ccmsipline”,nonce=“GBauADss2qoWr6k9y3hGGVDAqnLfoLk5”,
uri=“sip:172.18.197.224”,algorithm=MD5,response=“126c0643a4923359ab59d4f5349455
2e”
CUCM receives the registration message and searches for an endpoint that matches
the provided username in the SIP message (3rdpsip in the preceding example).
CUCM uses the digest credentials configured for that user to verify the keyed hash
(response=“126c0643a4923359ab59d4f53494552e in the preceding example).
CUCM searches for a third-party SIP phone that is associated with the end user and verifies
that the configured directory number matches the one provided by the third-party SIP phone
in its registration message. If the phone is found and the directory number is the same, the
third-party SIP phone registers with CUCM.
To add a third-party SIP phone in CUCM, follow these steps:
Step 1 Configure an end user in CUCM and specify the digest credentials.
NOTE CUCM must be explicitly configured to verify the keyed hash. By default,
CUCM searches only for the end-user name.
Review Questions 161
Step 2 Add the third-party SIP phone in CUCM and configure its directory
number.
Step 3 Associate the third-party SIP phone with the end user.
Step 4 Configure the third-party SIP phone with the IP address of the CUCM,
end-user ID, digest credentials, and directory number.

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