SIPPhoneReleaseNotes.2.2.pdf

Release Notes for Cisco SIP IP Phone 7940/7960

Release 2.2

November 6, 2001

Contents

This document lists the known problems in the Cisco SIP IP Phone 7940/7960 Version 2.2 and contains

information about the Cisco SIP IP Phone 7940/7960 (hereafter referred to as the Cisco SIP IP phone)

that was not included in the Cisco SIP IP phone documentation.

This document includes the following sections:

•

Contents, page 1

•

New and Changed Information, page 1

•

Caveats, page 3

•

Related Documentation, page 5

•

Obtaining Documentation, page 6

•

Obtaining Technical Assistance, page 7

New and Changed Information

For detailed information about each new feature and a list of all the Cisco SIP IP phone features, refer

to the Version 2.2 of the Cisco SIP IP Phone Administrator Guide at the following URL:

http://www.cisco.com/univercd/cc/td/doc/product/voice/c_ipphon/sip7960/sipadm22/index.htm

Corporate Headquarters:

Cisco Systems, Inc., 170 West Tasman Drive, San Jose, CA 95134-1706 USA

New and Changed Information

The following new features have been added to the Cisco SIP IP phone Version 2.2:

REFER Support

The Cisco SIP IP Phone uses the SIP REFER method to initiate a transfer. The REFER method allows

for the initial parties to be reconnected upon a failed attended transfer attempt. REFER is used if the

target endpoint supports the method; otherwise, the phone will failover to use the BYE/Also method of

transfer in earlier releases.

There are no configuration changes associated with transfer. The REFER method is used as the default.

The BYE/Also method of transfer is accepted also if received by the phone instead of a REFER.

7940 Support

The Cisco SIP IP Phone software recognizes the phone model (7960 or 7940) it is booting and provides

the correct 6-line (7960) or 2-line (7940) support. The Cisco SIP IP Phone 7940 data sheet is located at

the following URL:

http://www.cisco.com/warp/public/cc/pd/tlhw/prodlit/7940_ds.htm

OPTIONS Support

When an OPTIONS request is received, the Cisco SIP IP Phone responds with a list of phone methods

and parameters supported. The Cisco SIP IP Phone does not generate an OPTIONS request.

Configurable VoIP Control Port

The listen port for SIP messages is configurable. When the NAT enable flag is used in conjunction with

the VoIP Control Port, the packets are sourced from this port rather than from an ampherol port. See the

voip_control_port parameter descriptions in the section, “Modifying the Default SIP Configuration File”

in Chapter 3, “Managing Cisco SIP IP Phones,” at the following URL:

http://www.cisco.com/univercd/cc/td/doc/product/voice/c_ipphon/sip7960/sipadm22/index.htm

Configurable RTP Media Ports

The Cisco SIP IP Phone allows for the start and end RTP media ports to be configured. See the

start_media_port and end_media_port descriptions in the section, “Modifying the Default SIP

Configuration File” in Chapter 3, “Managing Cisco SIP IP Phones,” at the following URL:

http://www.cisco.com/univercd/cc/td/doc/product/voice/c_ipphon/sip7960/sipadm22/index.htm

NAT Support

When network address translation (NAT) is enabled, the Cisco SIP IP Phone provides support for SIP

messages to traverse NAT/Firewall networks. The Contact and Via headers are modified to reflect the

NAT parameters. The Cisco SIP IP Phone can also enable NAT received processing. See the nat_enable,

nat_address, and nat_received_processing parameters in the section, “Modifying the Default SIP

Configuration File” in Chapter 3, “Managing Cisco SIP IP Phones,” at the following URL:

http://www.cisco.com/univercd/cc/td/doc/product/voice/c_ipphon/sip7960/sipadm22/index.htm

Release Notes for Cisco SIP IP Phone 7940/7960 Release 2.2

Caveats

Outbound Proxy Support

When an Outbound Proxy is configured, all SIP requests are sent to the Outbound Proxy Server instead

of the configured Proxy Address. The Cisco SIP IP Phone does not have to register with the Outbound

Proxy. All interactions, such as authentication, with the Outbound Proxy are treated the same as the

interactions with the primary proxy. See the outbound_proxy and outbound_proxy_port descriptions in

the section, “Modifying the Default SIP Configuration File” in Chapter 3, “Managing Cisco SIP IP

Phones,” at the following URL:

http://www.cisco.com/univercd/cc/td/doc/product/voice/c_ipphon/sip7960/sipadm22/index.htm

Caveats

This section describes caveats, or known problems, that are open in this release. It also lists closed

caveats in this release.

Open Caveats - Release 2.2

CSCds27516— When the network media is manually configured, the inline power does not work

when connected to a Catalyst 3500 switch.

Problem Description: When the network media is manually configured, the inline power support

does not work when the Cisco SIP IP phone is connected to a Catalyst 3500 switch.

Recommended Action: When connecting to a Catalyst 3500 switch, configure the phone to

automatically negotiate the network media type by selecting Auto for the Network Media Type

parameter located in the Network Configuration menu.

•

CSCds35841: When in overview mode, the Cisco SIP IP phone soft keys do not work.

Problem Description: Pressing a line button during a call displays the overview screen on which

there is located a Redial and NewCall soft key. However, these soft keys are ignored by the phone

if pressed.

Recommended Action: Return to the call screen (wait 8 seconds for the call screen to reappear or

press the line button again).

•

CSCds64602: Caller cannot terminate a call transferred back by the Callee.

Problem Description: The Cisco SIP phone does not properly handle the following call scenario:

– Phone A calls phone B.

– Phone B performs a call transfer with consultation back to phone A.

– Phone B’s call hangs up correctly, however, the phone A’s call has no audio and requires several

on and off hooks to terminate the call.

Recommended Action: Press the speaker button or go off hook and back on hook several times to

terminate the call.

•

CSCdt89255: 7960 SDP Codec negotiation issue causes one-way voice.

Problem Description: The 7960 SIP IP Phone SDP codec negotiation can cause one-way voice

when other endpoint is using asynchronous codec support.

Recommended Action: Ensure other clients support single codec. The 7960 SIP IP Phone does not

support asynchronous codec.

•

Release Notes for Cisco SIP IP Phone 7940/7960 Release 2.2

Caveats

•

CSCdu35450: proxyN_port of UNPROVISIONED does not default to 5060.

Problem Description: The proxyN_port parameter in the SIP 7960 IP Phone does not default to

5060. It defaults to 0 instead.

Recommended Action: Set the proxyN_port parameter to 5060, rather than "" or

"UNPROVISIONED".

•

CSCdu43127: DSP Timeouts with multiple instances of DTMF and speakerphone.

Problem Description: On rare occasions when using speakerphone, G729a codec, VAD enabled,

and DTMF, the phone encounters a DSP timeout which affects only the active call. Any subsequent

calls work unless the above four factors are used again. The DSP timeout occurs only when all four

of the above factors happen simultaneously, and affects only the active call.

This issue is also a side effect of excessive debug use.

Recommended Actions:

–

Use the Handset instead of the speakerphone, or

–

Use the default codec G711ulaw, or

–

Use the default of VAD as disabled.

•

CSCdu43128: Invalid SRV in maddr of Contact causes hung midcall INVITE.

Problem Description: After a basic call is established and a mid-call INVITE is sent to the phone,

with an invalid SRV entry in the Contact: header, the phone leaves the call hung. The SIP messaging

is correct in that the mid-call INVITE gets rejected with a 500 Internal Server error, but the display

still shows a connected call.

Recommended Action: Press the EndCall softkey or hang up the phone.

•

CSCdu68098: Requested-By header in BYE message missing for Transfer.

Problem Description : When using the BYE/Also method of transfer the BYE message does not

include the Requested-By header.

Recommended Action : There is no workaround. However, the primary method for Transfer is now

REFER so the BYE/Also method is used only when an endpoint cannot support REFER.

•

CSCdu68091 : No support for configurable action tag in REGISTER Contact.

Problem Description : When the phone sends a REGISTER message it does not attach an

action= tag to the Contact header. This can lead to mismatched registrations when another client

REGISTERs with the same user ID because the 7960/7940 is always treated as action=none.

Recommended Action : Configure the other client to have action=none to avoid mismatched

registrations.

•

CSCdv33556: INVITEs with multiple valid m= lines get rejected with 400 Bad Request.

Problem Description : When the phone receives an INVITE with multiple valid m= SDP lines (for

example, m=application and m=audio), a 400 BAD Request is sent in response.

Recommended Action : None.

•

CSCdv26487 : CANCEL needs to handle Proxy challenges similar to BYE.

Problem Description : There is no support for a CANCEL with credentials. If a CANCEL gets

challenged by the Proxy, a new CANCEL with Proxy-Authorization credentials will not be sent.

Recommended Action : None; there is no workaround.

Release Notes for Cisco SIP IP Phone 7940/7960 Release 2.2

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