The SIP server is the main component of the IP PBX and is responsible for establishing all SIP phone calls in the network. It is also called a SIP proxy server or a registration server. Normally, the server does not participate in the media processing process. In SIP networks, media generally always adopts end-to-end negotiation processing. In some special cases or business processes, such as Music On Hold, the server will also actively participate in media negotiation. A simple SIP server is only responsible for session establishment, maintenance, and cleanup, but with multiple interfering calls. The relatively complex server, also known as SIP PBX, not only provides support for basic calls and basic sessions, but also provides rich services such as Presence, Find-me, Music On Hold, and so on. Most servers are based on the Linux platform, and the typical representatives are: Kamailio, OpenSER, sipXecx, etc. There are also some servers based on the Windows [2] platform, typically represented by: miniSipServer, Brekeke, etc.
UC features for up to 300 users
Our sip server supports up to 300 sip telephones at the same time, which means you can manage up to 300 phones in a single phone system.This secure and reliable IP PBX delivers unifed communication features at an unprecedented price point it hout any licensing fees, costs-per-feature, or recurring fees.The IP PBX system supports holding teleconference, and several meeting Bridges can be held at the same time. Inside or outside line Numbers can be invited during the meeting, and the meeting room can be recorded all the way.The Sip server is the main component of ip pbx and is responsible for establishing all sip phone calls on the network.Normally, the server does not participate in the media processing. Throughout the ip pbx, media is always handled in an end-to-end negotiation. It provided by KNTECH is responsible for session establishment, maintenance, and cleaning. It not only provides basic calls, basic session support, but also provides rich media functions such as queue, broadcast, and multi-party calls.
The default port number of the Sip protocol is 5060, which belongs to the udp protocol.
The 5060 registered port of the SIP can be modified to other unoccupied UDP ports. In particular, accessing the server through the public network and modifying the default port number can improve the security of the system.
Servers running the sip protocol can accept unsecure or secure connections. Secure connection, mutual TLS is used with SIP and RTP communication.
By default, the sip server listens for insecure communication on the 5060 TCP port.
Listen for secure communication on port 5061.
SIP sessions use up to four main components: SIP User Agent, SIP Registration Server, SIP Proxy Server, and SIP Redirect Server. These systems complete SIP sessions by transmitting messages that include the SDP protocol (which defines the content and characteristics of the message). SIP User Agents (UAs) are end-user devices, such as mobile phones, multimedia handheld devices, PCs, PDAs, etc., used to create and manage sip sessions. The user agent client sends a message. The user proxy server responds to the message.
Sip stands for session initiation protocol and is mainly used in VOIP but it is not a Voip protocol. Sip is an application layer protocol. Sip mainly replaces traditional signal carriers through the Internet for language and video functions. The sip protocol is an Internet-based text protocol, so it has more powerful functions, such as text transmission and multi-party conferencing.
Sip server can be combined with voip gateway, sip phone into ip pbx.It also called sip proxy, sip proxy server, mainly used for call processing in ippbx as a telephone, managing call setup, disconnection and control between sip phones routing. The server usually has basic functions such as proxy, redirection, registration, etc. The sip proxy server usually forwards the request and response received by one UA or another proxy to another proxy. SIP redirect server responds to sip request, but does not Request to transfer to another user. Sip registration server, you can register the sip phone to the server and record the information of the sip phone. Sip server software integrates these three functions into one software and is processed by one server.
1. Establish a session between two or more endpoints (multiple conference rooms use multiple points, sessions include voice and video)
2. Perform data transmission of text, voice and video.
3. Transfer the session
4. Create multiple users for free
5. Free calls
6.Web-style interface management
7. Terminate the session
KNTECH IP PABX Model Number simultaneous calls
Register user | Simultaneous Calls | Model Number |
50 line | 15 line | KNTD-50 |
100 line | 30 line | KNTD-100 |
300 line | 90 line | KNTD-300 |
500 line | 150 line | KNTD-500 |
800 line | 400 line | KNTD-800 |
1000 line | 300 line | KNTD-01K |
3000 line | 900 line | KNTD-03K |
5000 line | 1500 line | KNTD-05K |
Model |
KNTD-300 IPPBX Enterprise - class unified communication system |
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Hardware interface |
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Network Interface |
Using dual Gigabit interface (10/100 / 1000M Ethernet adaptive) and integrated POE (IEEE 802.3at-2009) The third Gigabit port for the hot backup cluster |
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NAT router |
Support (user configuration) |
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Reset switch |
Support, Web configuration |
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Audio / video capability |
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Speech algorithm |
Support fax tone detection and automatic switching G.711, 7.29, 7.23, 7.26 encoding |
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Voice |
G.711 A-law / U-law, G.722, G.723.1, G.726, G.729A / B, AAL2-G.726-32 |
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Video coding |
H.264, H.263, H263 + |
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QoS |
Layer 3 QoS, Layer 2 QoS |
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Signaling protocol |
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DTMF detection method |
Support three, In Audio, RFC2833, SIP INFO |
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Digital signal |
PRI, SS7, MFC / R2 |
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Terminal management and automatic deployment capabilities |
Support for SIP terminal plug-and-play auto-deployment (including DHCP Option 66 / multicast SIP SUBSCRIBE / MDNS Automatic detection and configuration), support TFTP / HTTP / HTTPS upgrade, local upgrade |
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Network protocol |
TCP / UDP / IP, RTP / RTCP, ICMP, ARP, DNS, DDNS, DHCP, NTP, TFTP, SSH, HTTP / HTTPS, SIP (RFC3261), STUN, SRTP, TLS / SIP, LDAP |
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PSTN disconnection method |
Support the PSTN line intelligent detection and automatic matching function, support five kinds of signaling detection methods (1, Busy Tone detection, 2, CPT detection, 3, polarity reversal detection, 4, Hook Flash Timing detection, 5, Loop Current Disconnect detection ) |
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Safety |
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Media encryption |
SRTP, TLS, HTTPS, SSH |
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Advanced defense |
Fail2ban, Alert events, whitelists, blacklists, encrypted access controls |
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Physical characteristics |
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Global universal power supply |
Input: 100 ~ ~ 240VAC, 50 / 60Hz; |
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weight |
Equipment weight: 2.165 Kg; Package weight: 3.012 Kg |
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Use the environment |
Operating environment: 32 - 113ºF / 0 ~ 45ºC, humidity 10 - 90% (No condensation) |
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size |
440mm (L) x 185mm (W) x 44mm (H) IU standard |
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installation |
Desktop or cabinet installation |
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Additional features |
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Multi-language support |
Web user interface supports English / Simplified Chinese, customizable IVR / voice prompts in English, Chinese, english English |
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Caller ID detection |
ETSI-FSK, ETSI-DTMF |
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Polarity reversal / Wink |
Support, option on / off control call setup and termination |
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Call Center |
Supports multiple call queues and queue queuing sounds |
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Customize Auto Attendant |
Supports Layer 3 IVR (Interactive Voice Response) |
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Concurrent call capability |
Up to 5,000 SIP terminals can be registered, up to 500 concurrent calls |
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Meeting room capacity |
Supports up to 10 voice conference rooms, up to 100 voice conference members |
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Call function |
Call Park, Call Forward, Call Forward, Do Not Disturb, DISA, ring Group, Pickup Group, blacklist, call / intercom, etc. |
The system built on server is widely used in industry, and its rich functional form is suitable for enterprise office, voice call center, voice intercom function and so on.
This product model:KNTD-300
KNTD-1200
KNTD-800