Administrator’s Guide SoundPoint IP / SoundStation IP

CONFIG_FILES=”phone1[MACADDRESS].cfg, sip.cfg” MISC_FILES=””

LOG FILE DIRECTORY=”” OVERRIDES_DIRECTORY=””

CONTACTS_DIRECTORY=”” LICENSE_DIRECTORY=””/>

If you have a requirement for separate application loads on different phones on the same boot server, you can modify the application that is loaded when each phone reboots. An example is below:

<?xml version=”1.0” standalone=”yes”?>

<!-- Default Master SIP Configuration File -->

<!-- edit and rename this file to <Ethernet-address>.cfg for each phone. -->

<!-- $RCSfile: 000000000000.cfg,v $ $Revision:$ -->

<APPLICATION APP_FILE_PATH=”[PHONE_PART_NUMBER].sip.ld” CONFIG_FILES=”phone1.cfg, sip.cfg” MISC_FILES=””

LOG FILE DIRECTORY=”” OVERRIDES_DIRECTORY=”” CONTACTS_DIRECTORY=”” LICENSE_DIRECTORY=””/>

You can also use the substitution strings PHONE_MODEL,

PHONE_PART_NUMBER, and PHONE_MAC_ADDRESS in the master configuration file. For more information, refer to Product, Model, and Part Number Mapping on page C-26.

You can also direct phone upgrades to a software image and configuration files based on the phone model number and part number. All XML attributes can be modified in this manner. An example is below:

<?xml version=”1.0” standalone=”yes”?>

<!-- Default Master SIP Configuration File -->

<!-- edit and rename this file to <Ethernet-address>.cfg for each phone. -->

<!-- $RCSfile: 000000000000.cfg,v $ $Revision:$ --> <APPLICATION APP_FILE_PATH=”sip.ld” CONFIG_FILES=”phone1.cfg, sip.cfg” MISC_FILES=”” LOG_FILE_DIRECTORY=””

OVERRIDES_DIRECTORY=””

CONTACTS_DIRECTORY=”” LICENSE_DIRECTORY=”” APP_FILE_PATH_SPIP300=”SPIP300.sip.ld” CONFIG_FILES_SPIP300=”phone1_SPIP300.cfg, sip_SPIP300.cfg” APP_FILE_PATH_SPIP500=”SPIP500.sip.ld” CONFIG_FILES_SPIP500=”phone1_SPIP500.cfg, sip_SPIP500.cfg” />

For more information, refer to “Technical Bulletin 35361: Overriding Parameters in Master Configuration File on SoundPoint IP Phones“ at http://www.polycom.com/usa/en/support/voice/soundpoint_ip/VoIP_T echnical_Bulletins_pub.html .

Application Configuration

The configuration file sip.cfg contains SIP protocol and core configuration settings that would typically apply to an entire installation and must be set before the phones will be operational, unless changed through the local web

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Polycom SIP 3.1 manual Application Configuration, CONFIGFILES=phone1MACADDRESS.cfg, sip.cfg MISCFILES=
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SIP 3.1 specifications

Polycom SIP 3.1 is an advanced session initiation protocol designed to enhance voice and video communication in various business environments. As a pivotal component of Polycom’s telecommunication solutions, SIP 3.1 offers several features and characteristics that cater to the evolving needs of modern enterprises, particularly those that rely on seamless and efficient communication.

One of the standout features of Polycom SIP 3.1 is its robust interoperability. This protocol supports a wide range of endpoints and platforms, allowing organizations to integrate their existing systems with new technologies effortlessly. This flexibility ensures that businesses can leverage their previous investments while upgrading to the latest communication tools.

Another key aspect of Polycom SIP 3.1 is its enhanced call management capabilities. The protocol facilitates efficient call handling, enabling users to manage multiple calls seamlessly. Features like call hold, transfer, and conferencing are streamlined, which enhances productivity by allowing for efficient collaboration among team members. Additionally, it is optimized for low latency and high-quality audio, ensuring that conversations are clear and free from disruptions.

Security is paramount in today’s digital landscape, and Polycom SIP 3.1 addresses this concern with advanced encryption standards. By integrating security features such as Transport Layer Security (TLS) and Secure Real-time Transport Protocol (SRTP), it protects sensitive communication from unauthorized access and ensures that data remains confidential throughout the call.

Polycom SIP 3.1 also boasts compatibility with various video codecs, making it a versatile choice for video conferencing. This compatibility ensures high-quality video streams, which is essential for effective visual communication in remote meetings. Furthermore, the support for the H.264 codec provides efficient bandwidth usage, making high-definition video conferencing accessible, even in varying network conditions.

Moreover, the protocol provides strong support for presence and instant messaging, which enhances real-time communication among users. This integration of voice, video, and messaging capabilities fosters a more connected and collaborative work environment, allowing teams to engage effectively regardless of their geographical locations.

In summary, Polycom SIP 3.1 stands out as a sophisticated solution tailored to meet the demands of modern business communication. With its emphasis on interoperability, call management, security, video quality, and real-time collaboration, it caters to companies of all sizes seeking to optimize their communication infrastructure and enhance productivity in the workplace. As businesses continue to navigate the complexities of digital communication, Polycom SIP 3.1 remains a compelling choice in the market.
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