Starting with Firmware Release 11.3(1), the Administration Guide is enhanced by integrating the information from the previous Provisioning Guide. The enhanced Administration Guide describes how to configure the phones with either the configuration file or the phone web interface.

Two additional guides are added: FAQ for Cisco IP Phones with Multiplatform Firmware and BroadSoft Quick Start Guide.

The complete documentation suite is:

• Cisco IP Conference Phone 7832 Multiplatform Phones Administration Guide

• Cisco IP Conference Phone 7832 Multiplatform Phones User Guide

• Cisco IP Conference Phone 7832 Multiplatform Phones Release Notes

• FAQ for Cisco IP Phones with Multiplatform Firmware

• BroadSoft Quick Start Guide (for BroadSoft platform only)

Refer to publications that are specific to your language and call control system. Navigate from the following documentation URL:

https://www.cisco.com/c/en/us/support/collaboration-endpoints/ip-phone-7800-series-multiplatform-firmware/tsd-products-support-series-home.html

You can specify the cipher suites that the phone TLS applications use. This feature lets you select which TLS cipher suites are enabled or disabled on the phone.

To specify the cipher suites, use the TLS Cipher List field on the Voice > System tab of the phone administration web page. The cipher list consists of one or multiple cipher suites. For multiple cipher suites, the suites are colon-delimited. When you specify a valid cipher list, the list applies to all the TLS applications on the phone. A valid cipher list must follow a specific format. For the cipher list formats, see https://www.openssl.org/docs/man1.0.2/man1/ciphers.html.

By default, no cipher list is specified in this field. The system regards an invalid list as a blank value. With a blank or an invalid list in this field, the cipher suites used vary with applications.

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Cisco IP Conference Phone 7832 Multiplatform Phones Administration Guide

You can enable auto provisioning of your multiplatform phones with this feature. Domain Name System Service (DNS SRV) records establish connections between a service and a hostname. When the phone looks for the location of the provisioning service, it first queries on the given DNS SRV domain name, then it queries for SRV records. The phone validates the records to confirm that the server is accessible. Then, it continues to the actual provisioning flow. Service providers can utilize this DNS SRV provisioning flow to provide auto provisioning.

DNS SRV bases the hostname validation on the certificate of the DHCP provided domain name. It’s important that all SRV records use a valid certificate containing the DHCP provided domain name.

The DNS SRV query includes the DHCP domain name in its construction as follows: _<servicename>._<transport>.<domainName>.

For example, _ciscoprov-https._tls.example.com, instructs the phone to do a lookup for example.com. The phone uses the hostname and port number that is retrieved from DNS SRV query to build the URL and download the initial configuration from it.

DNS SRV is one of many auto provisioning mechanisms that the phone uses. The phone tries the mechanisms in the following order:

1. DHCP

2. DNS SRV

3. EDOS

4. GDS (Activation Code Onboarding), or EDOS Device Activation

DNS SRV doesn't support the TFTP protocol. If you use TFTP, you receive the following error message: Error - TFTP Scheme not supported in SRV lookups.

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Cisco IP Conference Phone 7832 Multiplatform Phones Administration Guide

Your phones now support RFC4733. You can choose from three audio-video transport (AVT) options for sending DTMF pulses to the server.

• AVT Dynamic Payload for 8 kHz clock rate

• AVT 16kHz Dynamic Payload for 16 kHz clock rate

• AVT 48kHz Dynamic Payload for 48 kHz clock rate

Configured dynamic payloads are used for outbound calls only when the phone presents a Session Description Protocol (SDP) offer.

You can configure the AVT Dynamic Payload options in the SDP Payload section of the phone web page.

Where to Find More Information

Cisco IP Conference Phone 7832 Multiplatform Phones Administration Guide

You can provide users with an activation code to set up their phones with this feature. Users enter the short activation code on the phone Welcome screen. Then, the phone receives its provisioning information.

This feature uses the cloud provisioning method via the CDA (EDOS) server. It doesn’t require MAC addresses added to the CDA server.

The short activation code has any number of digits between 3 and 16, inclusive. Users press pound (#), then enter the short code digits. Use this format: #nnn, #nnnnnnnnnnnnnnnn, or any number of digits between 3 and 16. The pound (#) key signals that the user is attempting to use Auto Provisioning with Short Activation Code. Otherwise, the phone expects the 16-digit code for the Activation Code Onboarding feature. If the user fails to enter pound (#) before the digits, the phone displays Invalid activation code. The same message displays if the user enters any wrong code.

If the phone fails to register with the correct code, reset the phone to the factory settings, and try again.

Where to Find More Information

• Cisco IP Conference Phone 7832 Multiplatform Phones Administration Guide

• Cisco IP Conference Phone 7832 Multiplatform Phones User Guide

You can enable the phone to send end-of-call statistics in Session Initiation Protocol (SIP) messages. The phone sends call statistics to a remote end when a call terminates or when a call is on hold. The call statistics are sent as headers in BYE and re-INVITE messages: RTP-RxStat and RTP-TxStat for audio sessions.

By default, the phone doesn't send call statistics. You can enable this feature using the Call Statistics field on the Voice > SIP tab of the phone administration web page.

Where to Find More Information

Cisco IP Conference Phone 7832 Multiplatform Phones Administration Guide

You can enable increased phone security on a phone line if you use TLS. The phone line can verify the hostname to determine if the connection is secure.

Over a TLS connection, the phone can verify the hostname to check the server identity. The phone can check both the Subject Alternative Name (SAN) and the Subject Common Name (CN). If the hostname on the valid certificate matches the hostname that is used to communicate with the server, the TLS connection establishes. Otherwise, the TLS connection fails.

The phone always verifies the hostname for the following applications:

• LDAPS

• XMPP

• Image upgrade over HTTPS

• XSI over HTTPS

• File download over HTTPS

• TR-069

When a phone line transports SIP messages over TLS, you can control the hostname verification with the TLS Name Validate field on the Ext(n) tab. Set the field to Yes to verify the hostname. Set the field to No to bypass the hostname verification.

Where to Find More Information

Cisco IP Conference Phone 7832 Multiplatform Phones Administration Guide

You can choose client-initiated mode or server-initiated mode for media plane security negotiations between the phone and the server. Set the MediaSec Request field to Yes to enable the phone to initiate negotiations with the server for media plane security. The default setting is No (server-initiated mode). The security mechanism follows the standards stated in RFC 3329 and its extension draft Security Mechanism Names for Media (See https://tools.ietf.org/html/draft-dawes-sipcore-mediasec-parameter-08#ref-2). You can use the MediaSec Over TLS Only field to limit that media plane security negotiation is applied only when SIP signaling transport protocol is TLS.

Access the parameters in the SIP Settings section on the Voice > Ext (n) tab of the phone web interface.

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Cisco IP Conference Phone 7832 Multiplatform Phones Administration Guide

We have improved phone security by hardening the operating system. Hardening ensures that the phone has a firewall to protect it from malicious incoming traffic.

The firewall tracks the ports for incoming and outgoing data. It detects incoming traffic from unexpected sources and blocks the access. The firewall allows all outgoing traffic.

The firewall is enabled by default, and is configured with the default open UDP and TCP ports. You can change the setting from the Voice > System > Security Settings > Firewall section of the phone web page. Firewall changes don't require the phone to reset. Phone soft restarts generally don't affect firewall operation.

You can also control other firewall options with keywords. For more information, see the administration guide.

Where to Find More Information

Cisco IP Conference Phone 7832 Multiplatform Phones Administration Guide

You can set up multicast paging to allow the users to page to phones. The page can go to all phones or a group of phones in the same network. Any phone in the group can initiate a multicast paging session. Only the phones that listen for the paging group can receive the page.

You can add a phone to a maximum of ten paging groups. Each paging group has a unique multicast port and number. The phones within a paging group must subscribe to the same multicast IP address, port, and multicast number.

You can configure the priority for the page from a specific group. When a phone is active and a page session with a higher priority occurs, the user hears the page on the active audio path. Configure the priority on the Voice > Phone tab of the administration web page.

When multiple paging sessions occur, the phone answers them in chronological order. The phone doesn't answer the next page until the active page ends. When do not disturb (DND) is enabled, the phone ignores incoming paging.

You can specify a codec for the paging to use. The supported codecs are G711a, G711u, G722, and G729. If you don't specify the codec, paging uses G711u by default.

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Cisco IP Conference Phone 7832 Multiplatform Phones Administration Guide

The multiplatform phones now support OPUS Codec Narrowband. The OPUS Codec Narrowband saves on network bandwidth. The multiplatform phones support OPUS codec, and as of Firmware Release 11.3(1), you can configure the phone to use both wideband and narrowband. The phones can use the OPUS narrowband and wideband codecs without conflict.

Where to Find More Information

Cisco IP Conference Phone 7832 Multiplatform Phones Administration Guide

You can configure programmable softkeys (PSK) with dual tone multifrequency (DTMF). This configuration enables the phone to send digital pulses to the server during an active call. When you enable a function on a PSK, the user sees the softkey name, and presses it to perform the named function.

This feature applies only to programmable softkeys. It doesn’t apply to programmable line keys (PLK). If you configure any PLK for this feature, the display presents the Circled X icon Ⓧ, and nothing happens when you press the key.

This feature supports only Connected Key List and Connected Video Key List.

Where to Find More Information

Cisco IP Conference Phone 7832 Multiplatform Phones Administration Guide

You can configure your phone to start an RTP session either before or after the phone receives an ACK from the calling party. This configuration optimizes the media transmission in an environment that experiences audio losses caused by the ACK transmission delay.

Use the RTP Before ACK field on the Voice > SIP tab of the phone administration web page. When set to Yes, the RTP transmission doesn't await an ACK, but starts after a 200 OK message is sent out. When set to No, the RTP transmission doesn't start until an ACK is received. The default setting is No.

Where to Find More Information

Cisco IP Conference Phone 7832 Multiplatform Phones Administration Guide

The phones now support “Session Identifier”. This feature helps to overcome the limitations with the existing call-identifiers and allows end-to-end tracking of a SIP session in IP-based multimedia communication systems in compliance with RFC 7989. To support session identifier, “Session-ID” header is added in the SIP request and response messages.

You can enable this feature from Voice > Ext(n) > SIP Settings > SIP SessionID Supports in the phone web interface.

Where to Find More Information

Cisco IP Conference Phone 7832 Multiplatform Phones Administration Guide

The phones display the complete Unique Device Identifier (UDI) information on the phone screen and on the phone web page. Cisco UDI provides a unique identity for all Cisco hardware products on which it has been enabled. The UDI is composed of three data elements associated with the phone. The data elements are:

• Product Identifier (PID)

• Version Identifier (VID)

• Serial Number (SN)

To view the UID information, do one of the following:

• On your phone, select Settings > Status > Product information

• In the phone web interface, select Info > Status >Product Information.

Where to Find More Information

• Cisco IP Conference Phone 7832 Multiplatform Phones Administration Guide

• Cisco IP Conference Phone 7832 Multiplatform Phones User Guide

You can assign a name to your Voice Quality Metrics (VQM) SIP Publish Message report. The name helps you organize your voice quality reports.

You add your report name in the Voice Quality Report Group field on the Voice > Ext(n) tab of the phone web page. Your report name can't begin with a hyphen (-), semicolon (;), or a space.

You can also configure the Voice Quality Report Group field in the XML configuration file.

You can look up your report in a packet analyzer, such as, Wireshark. In the packet analyzer, the Voice Quality Report Group field appears as LocalGroup.

Where to Find More Information

Cisco IP Conference Phone 7832 Multiplatform Phones Administration Guide

The following list contains the severity 1, 2, and 3 defects that are open for the Cisco IP Phone 7800 Series Multiplatform Phones that use Firmware Release 11.3(1).

For more information about an individual defect, you can access the online history for the defect by accessing the Bug Search tool and entering the Identifier (CSCxxnnnnn). You must be a registered cisco.com user to access this defect information.

Because the defect status continually changes, the list reflects a snapshot of the defects that were open at the time this report was compiled. For an updated view of the open defects or to view specific bugs, access the Bug Search Toolkit as described in View Caveats.

• CSCvr58405 CP-7821-3PCC: Directory list via XML is not searchable

• CSCvr76623 CP-7841-3PCC-K9= Conference URI not working

• CSCvr14202 Phone would not register after switch back from static ipv6 to DHCP ipv6

• CSCvr90430 7832 8832 LED status is incorrectly displayed on web page

The following list contains the severity 1, 2, and 3 defects that are resolved for the Cisco IP Phone 7800 Series Multiplatform Phones that use Firmware Release 11.3(1).

For more information about an individual defect, you can access the online history for the defect by accessing the Bug Search tool and entering the Identifier (CSCxxnnnnn). You must be a registered cisco.com user to access this defect information.

Because the defect status continually changes, the list reflects a snapshot of the defects that were resolved at the time this report was compiled. For an updated view of the resolved defects or to view specific bugs, access the Bug Search Toolkit as described in View Caveats.

• CSCvr58190 Linux Kernel allocate_trace_buffer Function Double Free Vulnerability

• CSCvr67582 Linux Kernel register_queue_kobjects() Denial of Service Vulnerability

• CSCvq85320 3PCC-7841: Reverse Name Lookup does not work

• CSCvq94980 Multiple Vulnerabilities in curl

• CSCvn38040 Choppy audio during early media

• CSCvm70118 From missed call screen no way to go back

• CSCvs02868 1-way audio on OPUS codec if remote does not send OPUS codec fmt