Index

missed call configuration A–112 Network Address Translation A–120 no answer A–115

quotas A–94 registration A–107 roaming buddies A–122 roaming privacy A–123 routing A–118 routing server A–118

per-phone configuration file A–106 phone diagnostics 5–9 phone1.cfg A–106

port <port> A–62 presence 4–60 presence <pres> A–72

product-model-part number mapping C–26 protocol <voIpProt> A–6

protocol server <server> A–7

protocol special events <specialEvent> A–16 provisioning <prov> A–90

provisioning over CLink 4–39 provisioning protocols 3–4 provisioning protocols, supported 3–4

Q

QOS. See also Quality of Service Quality of Service <QOS> A–55 quotas <quotas> A–94

R

RAM disk <ramdisk> A–90 rebooting phones 3–17,3–20 receive equalization <rxEq> A–49 registration <reg> A–107

reliability of provisional responses B–9 request <request> A–91

request delay <delay> A–91

request validation <requestValidation> A–15 resetting to factory defaults 3–5

resource <res> A–93 resource files, overview 2–7 RFC support B–2

ring type <rt> A–36 ringer patterns A–34

roaming buddies <roaming_buddies> A–122 roaming privacy <roaming_provacy> A–123 routing <routing> A–118

routing server <server> A–21,A–118

RTP <RTP> A–56,A–57,A–62

S

sampled audio files A–31

sampled audio for sound effects <saf> A–30 SCA. See also shared call appearances scheduled logging parameters A–87

SDP <SDP> A–9

secure real-time transport protocol 4–82 security <sec> A–88

server menu 3–9

server redundancy 4–56

server-based call forwarding See also call forwarding

server-based DND See also do not disturb Services key. See also Applications key Session Initiation Protocol

setting up

advanced features 4–22 audio features 4–73 basic features 4–1 boot server 3–12 network 3–2 security features 4–80

shared call appearance signaling B–10 shared call appearances

shared calls <shared> A–67

shared lines barge-in 4–27,A–109

SIP

1xx Responses - Provisional B–6 2xx Responses - Success B–6 3xx Responses - Redirection B–7 4xx Responses - Request Failure B–7 5xx Responses - Server Failure B–8 6xx Responses - Global Failure B–8 application architecture 2–3

basic protocols, hold implementation B–9 basic protocols, request support B–3 basic protocols, response support B–6 basic protocols, RFC and Internet draft

support B–2

basic protocols, transfer B–9

instant messaging and presence leveraging extensions B–10

RFC 2–1

SIP application description 2–4 installing 3–14 upgrading 3–19

Index – 5

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Polycom SIP 3.1 manual SDP SDP A-9

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.