The DINSTAR SBC3000 is designed to deliver security, interoperability, and transcoding for service providers and telecom operators VoIP interconnections. It is scalable from 500 to 2000 SIP sessions and offers High Availability (HA) with 1+1 active/standby redundancy, comprehensive SIP mediation, and intensive protection to ensure uninterrupted communications and services.
The DINSTAR SBC8000 is a software-based SBC designed to deliver robust security, seamless connectivity, advanced transcoding, and media controls to VoIP networks. It offers users the flexibility to deploy the SBCs on their dedicated servers, virtual machines, and private or public cloud, and to scale easily on demand. It supports up to 10,000 concurrent call sessions, 5,000 media transcoding, and 100,000 SIP registrations.
DINSTAR Session Border Controllers provide enhanced security features including protection against malicious attacks like DoS/DDoS, malformed packets, SIP/RTP flooding, perimeter defense against eavesdropping, fraud and service theft, TLS/SRTP for call security, topology hiding against network exposure, and ACL, Dynamic white & black list.
DINSTAR Session Border Controllers ensure easy management through an intuitive web interface, SNMP, remote web and telnet, configuration backup & restore, CDR report and export, and debug tools, statistics, and reports.
The DINSTAR SBC8000 Session Border Controller can handle up to 10,000 concurrent call sessions, 5,000 media transcoding, and 100,000 SIP registrations.
A Session Border Controller (SBC) is a network element deployed to protect SIP based VoIP networks. It provides security, interoperability, and transcoding for service providers and telecom operators VoIP interconnections.
Yes, the DINSTAR SBC8000 Session Border Controller offers users the flexibility to deploy the SBCs on their dedicated servers, virtual machines, and private or public cloud.
SIP (Session Initiation Protocol) is used in the DINSTAR Session Border Controller for setting up, modifying, and terminating multimedia sessions including VoIP calls. It provides interoperability and seamless connectivity in VoIP networks.
Yes, the DINSTAR Session Border Controller supports High Availability (HA) with 1+1 active/standby redundancy to ensure uninterrupted communications and services.
Transcoding in the DINSTAR Session Border Controller is used for media controls. It converts media streams from one codec to another to ensure seamless connectivity in VoIP networks.
A GSM VoIP Gateway is a device that enables direct routing between IP, digital, analog and GSM networks. With these devices, businesses can significantly reduce the money they spend on telephony, especially the costly calls from IP to GSM. The core idea behind cost saving with VoIP GSM Gateways is Least Cost Routing (LCR).
Dinstar GSM VoIP Gateway offers complete choices scalable from 1 to 64 channels, implement the state-of-the-art functionality in market as always, and enable the smooth transit between mobile and VoIP networks. Integrated GSM/WCDMA/LTE connectivity and SIP protocol compatible with mainstream VoIP platforms.
It is suitable for enterprises, multi-site organizations, call terminators and areas with limited landline like rural area to cut down telephony costs and enable easy & efficient communications.
Dinstar GSM VoIP Gateway has good compatibility with different vendors.
The Dinstar UC2000 series VoIP GSM/3G/4G Gateways offer complete choices scalable from 1 to 64 channels.
SIMBank & SIMCloud is a feature in Dinstar GSM VoIP Gateway that provides scalable options for users. However, specific details about this feature are not provided on the website.
UC2000-VH(EOL) is a model of Dinstar GSM VoIP Gateway. The term EOL stands for End of Life, indicating that this particular model may no longer be in production or supported.
UC2000-VA(EOL) is another model of Dinstar GSM VoIP Gateway. Similar to UC2000-VH(EOL), the term EOL indicates that this model may no longer be in production or supported.
Dinstar GSM VoIP Gateway implements state-of-the-art functionality to enable smooth transit between mobile and VoIP networks. It integrates GSM/WCDMA/LTE connectivity and is compatible with mainstream VoIP platforms.
Dinstar GSM VoIP Gateway helps in cost reduction by enabling direct routing between IP, digital, analog and GSM networks. This significantly reduces the cost of calls from IP to GSM.
An Analog VoIP Gateway is a device that connects legacy telephony equipment, such as analog phones and fax machines, to IP networks. It is used to transition from traditional telephony to IP-based systems without the need to replace existing telephony equipment.
Dinstar Analog VoIP Gateway offers a range of features including support for 1-128 FXS ports, fax over T.38 and pass-through, automatic firmware upgrade and configuration via HTTP/HTTPS, system logs and CDR, support for IPv6 and IPv4, and compatibility with SIP/IMS.
Telecom operators, service providers, enterprises, call centers, SMEs, and branch offices can benefit from using Dinstar Analog VoIP Gateway. It allows these organizations to leverage their existing telephony infrastructure while transitioning to IP-based systems.
Dinstar Analog VoIP Gateway is compatible with mainstream VoIP platforms and supports both IPv6 and IPv4.
The FXS VoIP Gateway is a type of Analog VoIP Gateway that supports 1-128 FXS ports. It allows the connection of analog phones, fax machines, and POS machines.
The FXO VoIP Gateway is another type of Analog VoIP Gateway that supports 4-32 FXO ports. It provides features such as support for fax over T.38 and pass-through, caller ID, polarity reserve, and automatic firmware upgrade and configuration via HTTP/HTTPS.
The Hybrid VoIP Gateway is a type of Analog VoIP Gateway that supports both FXS and FXO ports. It offers configurations such as 4 FXS 4 FXO and 8 FXS 8 FXO. It also supports power failure bypass.
Dinstar Analog VoIP Gateway supports fax machines through the T.38 protocol and pass-through. This allows fax data to be sent over IP networks.
Dinstar Analog VoIP Gateway is designed to work with both IPv6 and IPv4. This ensures that it can operate effectively on modern IP networks, regardless of the IP version used.
Dinstar Analog VoIP Gateway simplifies VoIP migration by allowing organizations to connect their existing analog phones, fax machines, and legacy PBX systems to IP PBX systems and IP Phone networks. This means that organizations can transition to IP-based systems without the need to replace their existing telephony equipment.
A Digital VoIP Gateway is a device that converts voice traffic into data packets for transmission over the internet. It is used to connect traditional telephony networks to IP networks.
Using a Digital VoIP Gateway can reduce costs, improve efficiency, and provide better voice quality. It also allows for easy scalability and integration with existing telephony systems.
A Digital VoIP Gateway works by converting analog voice signals into digital signals. These digital signals are then compressed and divided into packets that can be transmitted over an IP network.
Some key features to look for in a Digital VoIP Gateway include voice compression, echo cancellation, call routing, and security features. It should also be compatible with your existing telephony system.
Installation and configuration of a Digital VoIP Gateway can vary depending on the specific model and your network setup. It typically involves connecting the gateway to your network, configuring settings through a web interface, and setting up routing rules.
The main difference between a Digital VoIP Gateway and an Analog VoIP Gateway is the type of telephony system they connect to. A Digital VoIP Gateway is used to connect digital telephony equipment, while an Analog VoIP Gateway is used to connect analog devices.
Yes, a Digital VoIP Gateway can be used with your existing telephone system. It is designed to integrate with traditional telephony systems and convert voice traffic for transmission over an IP network.
The MTG1000 Series Digital VoIP Gateway is a product from DINSTAR that provides an easy and cost-effective way to transition from traditional telephony networks to IP-based networks. It supports a variety of telephony interfaces and offers high-quality voice communication.
The MTG200 Series Digital VoIP Gateway is a compact, cost-effective product from DINSTAR that is designed for small businesses. It supports a variety of telephony interfaces and provides high-quality voice communication.
You can buy a Digital VoIP Gateway directly from the DINSTAR website or from authorized DINSTAR distributors and resellers.
The A810 features a 10.1-inch IPS multi touch screen, HD voice quality, 4 SIP accounts, Wi-Fi, 6-way conference, and Bluetooth.
Yes, the A810 supports both ethernet and WiFi connect.
The A810 runs on Android 7.1 operation system.
The C61S/C61SP features HD voice quality, 2 SIP accounts, 2 line keys, 2.3”Graphic LCD, 5-way conference, and PoE.
The C61S/C61SP supports 2 SIP accounts.
Yes, the C61S/C61SP supports 5-way conference calls.
The C66G/C66GP features HD voice quality, 20 SIP accounts, 50 line keys, 4.3”Graphic LCD, 6-way conference, and Wi-Fi & Bluetooth Dongle.
The C66G/C66GP supports 20 SIP accounts.
Yes, the C66G/C66GP supports 6-way conference calls.
The C66G/C66GP has a 4.3”Graphic LCD.
The SIP Intercom products support H.264 video compression format and deliver excellent video quality in 720p video resolutions. For instance, the DP92V-SG model provides the same HD audio & video quality as wire cable type.
The door can be opened remotely, but also locally using a password or IC/ID card if there is an electronic door lock. This feature is available in models like the DP92V-SG and DP88.
The operating temperature range of the SIP Intercom products is -20 ℃ to 65 ℃. This is applicable for models like DPA and DP92V-SG.
The SIP Intercom products support two-way audio stream and HD voice. They also have a powerful echo cancellation function. This is applicable for models like DP92V-SG and DP88.
Yes, the SIP Intercom products support auto provisioning via FTP/TFTP/HTTP/HTTPS/PnP. This is applicable for models like DP92V-SG and DP88.
The SIP Intercom products come with a 2M Pixels color CMOS camera. For instance, the DP88 model supports up to 1280 x 720 resolution.
Yes, the SIP Intercom products support SIP over TLS and SRTP. This is applicable for models like DP92V-SG and DP88.
The SIP Intercom products support various codecs including PCMA, PCMU, G.729, G723_53, G723_63, G726_32, and wideband codec G.722. This is applicable for models like DP92V-SG and DP88.
The maximum image transfer rate of the SIP Intercom products is 1080P -30fps. This is applicable for models like DP88.
Yes, the SIP Intercom products support remote control through Action URL/Active URI. This is applicable for models like DP92V-SG and DP88.
The DINSTAR UC100 supports up to 100 extensions and 30 concurrent calls.
The DINSTAR UC2500 supports up to 500 extensions and 120 concurrent calls. It also offers features like voicemail, call recording, and VPN support.
Yes, the DINSTAR UC1500 is based on the X86 platform, allowing users to install third-party PBX software with simple installation.
The DINSTAR UC8000 supports up to 800 SIP extensions and 300 concurrent calls.
The DINSTAR UC350 supports 1000 extensions and 200 concurrent calls.
Yes, the DINSTAR UC1500 offers high reliability with redundant power supplies and hot swappable interface boards.
The DINSTAR UC2500 can handle up to 120 concurrent calls.
Yes, the DINSTAR UC8000 supports IP/SIP failover for enhanced reliability.
Yes, the DINSTAR UC1500 is based on the X86 platform, which allows for the installation of third-party IP PBX software such as Asterisk, Freeswitch, 3CX, Issabel, and VitalPBX.
The DINSTAR UC1500 supports FXS/FXO/E1/T1/LTE/GSM interfaces.
The customer, a Singapore Medical Group, operates 20 clinics across the island, located in various shopping malls. During the Covid-19 pandemic, the clinics faced a challenge maintaining social distancing norms as patients could not wait inside the clinic for their turn. If patients waited outside, they could not hear the receptionist call their turn. To address this issue, we designed an SMS Queue system with SMS and Voice Reminders.
In the world of communications, Voice over Internet Protocol (VoIP) has made a profound impact since its inception. From its early days in the late 1990s to the robust, reliable, and versatile technology we have today, VoIP has evolved significantly. With over a decade of experience designing and architecting VoIP projects, I've observed the transformative power of this technology, and I am excited to discuss what the future might hold for VoIP.
The Voice over Internet Protocol (VoIP) market in Southeast Asia is experiencing a significant surge in demand. This trend is driven by various factors, including technological advancements, increasing internet penetration, the need for business process optimization, and the cost efficiency of VoIP solutions.
In the current digital landscape, the security of Voice over Internet Protocol (VoIP) systems is of paramount importance. As businesses increasingly rely on VoIP for their communication needs, the potential for security breaches has also risen. While conventional firewalls are effective at protecting data networks, they are not equipped to handle the unique security challenges posed by VoIP systems. This necessitates the implementation of specialized security measures for VoIP systems.
Private Branch Exchange (PBX) systems have been a cornerstone of business communication for many years. However, with the advent of internet technology, a new player has entered the field: IP PBX. This article will explore the key differences between traditional PBX and IP PBX systems, helping you understand which might be the best fit for your business.
Voice over Internet Protocol (VoIP) technology has revolutionized the way we communicate, offering more flexibility, better sound quality, and cost savings compared to traditional telephony systems. One area where VoIP has made a significant impact is in the realm of intercom systems.
In the modern digital workspace, voice over Internet Protocol (VoIP) systems have become an integral part of business communications. However, as with any internet-connected technology, they are also vulnerable to cyber threats. This is where DINSTAR's Session Border Controllers (SBCs) come into play. These network elements are specifically designed to safeguard SIP-based VoIP networks and ensure seamless real-time communications.
The client is a Singapore-based Limosine Company that provides first-class service to VIP customers of several banks' private banking departments. The company faced a challenge where limosine services were often requested outside office hours, and sometimes drivers or customers overlooked the booked appointments, causing issues between the drivers and customers. To resolve this problem, we designed an SMS Limosine Schedule system with SMS and Voice Reminders.
Our client, a Singapore-based Tuition School, operates 12 tuition centres across the island. They cater to primary school students, offering classes in Chinese, Math, Science, and English. Each centre has 6 to 8 small classrooms, with an average of 10 students per class.The central management office was responsible for creating the class schedule for all the centres, including tutor assignment, student allocation, class scheduling and rescheduling, class merging, class transfers, and replacement...
Our customer is a multinational manufacturing & sales company based in Singapore, with additional offices in Taiwan, Australia, and Cambodia. Employing over 100 office staff and 500 factory workers, the customer required a secure and reliable communication network.