Dtmf Tone Generation, Dtmf Event RTP Payload

Configuring Your System

packets (also known as Silence Insertion Descriptor (SID) frames) and also decodes CN packets, efficiently regenerating a facsimile of the background noise at the remote end.

Configuration changes can performed centrally at the boot server:

Central (boot server)

Configuration file: sip.cfg

Enable or disable VAD and set the detection threshold.

•For more information, refer to Voice Activity Detection <vad/> on page A-52.

DTMF Tone Generation

The phone generates dual tone multi-frequency (DTMF) tones in response to user dialing on the dial pad. These tones are transmitted in the real-time transport protocol (RTP) streams of connected calls. The phone can encode the DTMF tones using the active voice codec or using RFC 2833 compatible encoding. The coding format decision is based on the capabilities of the remote end point.

Configuration changes can performed centrally at the boot server:

Central	Configuration file:	Set the DTMF tone levels, autodialing on and off times, and other
(boot server)	sip.cfg	parameters.
		• For more information, refer to Dual Tone Multi-Frequency
		<DTMF/> on page A-28.

DTMF Event RTP Payload

The phone is compatible with RFC 2833 - RTP Payload for DTMF Digits, Telephony Tones, and Telephony Signals. RFC 2833 describes a standard RTP-compatible technique for conveying DTMF dialing and other telephony events over an RTP media stream. The phone generates RFC 2833 (DTMF only) events but does not regenerate, nor otherwise use, DTMF events received from the remote end of the call.

Configuration changes can performed centrally at the boot server:

Central	Configuration file:	Enable or disable RFC 2833 support in SDP offers and specify the
(boot server)	sip.cfg	payload value to use in SDP offers.
		• For more information, refer to Dual Tone Multi-Frequency
		<DTMF/> on page A-28.

Acoustic Echo Cancellation

The phone employs advanced acoustic echo cancellation (AEC) for hands-free operation. Both linear and non-linear techniques are employed to aggressively reduce echo yet provide for natural full-duplex communication patterns.

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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.

Configuring Your System

Central (boot server)

DTMF Tone Generation

Central

<DTMF/> on page A-28.

DTMF Event RTP Payload

Central

Acoustic Echo Cancellation

SIP 3.1 specifications