PortSIP PBX provides extended security options, advanced privacy features, and built-in compliance options for industry and regional requirements so you can communicate and collaborate securely.

Data privacy and security processes

PortSIP offers a robust communications solution designed with a primary focus on keeping customer data secure. This commitment to privacy and security is reflected in every aspect of our work, from the design and development stages to the deployment and maintenance of our networks, platforms, and applications. PortSIP employs multiple technologies, procedures, and teams to ensure the communication solution meets privacy and security requirements.

Recording privacy and security processes

To comply with various countries’ regulations regarding call recording, PortSIP PBX offers the following recording features:

1. System Administrators can control whether to record calls between two external callers. For instance, when Caller A calls a contact center agent, and the agent subsequently refers Caller A to Caller B’s PSTN phone number, the conversation occurs between Caller A and Caller B. PortSIP PBX, based on the system administrator’s configuration, can automatically stop recording their conversation.
2. Tenant Administrators can configure a prompt to play when users call into PortSIP PBX. If the user does not agree to the recording, the system can automatically disconnect the call.

IP Phone security

Typically, most PBX systems store an extension's IP phone auto-provisioning file in a fixed folder path. Unfortunately, this approach can lead to security issues. For instance, if someone gains knowledge of another person's IP phone MAC address, they can easily download the configuration file to get the extension's password.

To address this security vulnerability, PortSIP PBX takes a proactive approach. Specifically, each extension's IP phone configuration file is stored in a separate directory with a randomized name. This measure ensures that even if an attacker obtains the phone's MAC address, they won't be able to guess the configuration file's download URL easily.

User provisioning

Authentication and authorization

PortSIP PBX employs standard methods to ensure secure authentication and authorization for users, whether they represent a small business or a federal government agency requiring the utmost level of security. If the organization utilizes a username and password system, PortSIP PBX offers a customizable password policy. This allows customers to enhance password entropy. Customers can modify the complexity of their passwords by adjusting the password policy, which may include the number of required characters, the use of special characters, uppercase letters, and numbers.

Additionally, PortSIP PBX supports Single Sign-On (SSO) with Microsoft 365. This feature enables enterprises to redirect users from PortSIP to Microsoft 365, allowing them to utilize the password and authentication flows of Microsoft 365.

Password security

After creating a user with the Admin, Standard User, or Standard International User role, an extension user was created, there are two passwords with a user.

• SIP Password. It's used for the IP Phone, Softphone, and WebRTC client to register to PortSIP PBX.
• User Password. It's used for the user to sign the PBX Web Portal to check voicemail, recording, CDR
• Both SIP Password and User Password must meet the tenant's password policy.

Login Security

There are some settings that allow the PBX system administrator to control login security for tenant managers, tenants, and extensions.

• Set the maximum number of login tries on the Web Login page, and the user IP will be blocked if the number of failed login attempts exceeds the allowed times.
• Set the period of an IP block, a blocked IP will be removed after this time.

Flexible admin control for security

You can assign a role type for a user when creating it and change it later as well. There are three roles by default: Admin, Standard User, and Standard International User. The Admin user has all permissions in the tenant scope and can manage the whole tenant. The Standard International User has permission to make calls to local, national, and international numbers. The Standard User only has permission to make calls between users.

Administrator security control functions

• Enable or disable an extension
• Modify roles to manage permissions for both national and international calls
• Adjust the password policy
• Restrict calls to specific countries
• Block calls based on specific prefixes of dialed numbers

SIP and TCP/IP security

PortSIP PBX offers security features primarily designed to thwart malicious attacks aimed at the PortSIP PBX, particularly if the administrator hasn’t implemented necessary precautions at the firewall level. It operates by identifying and blocking packet floods, DoS attacks, or brute force dictionary attacks that attempt to decipher the extension number and password. This includes:

• Protection against failed SIP authentication
• Handling of failed SIP authentication challenge requests (407)
• Control of TCP and UDP packet rates

User-Agent blacklist

To safeguard against malicious activities such as SPIT (Spam over Internet Telephony), TDoS (Telephony Denial-Of-Service), fuzzing, and War dialing, PortSIP PBX offers a feature that blocks specific User-Agents found in SIP messages. This feature is instrumental in enhancing the security of your telecommunication services.

Whitelist/Blacklist

PortSIP PBX allows you to whitelist and blacklist IP addresses. All traffic originating from whitelisted IP addresses will be allowed unchecked by the anti-hacking features. All traffic originating from blacklisted IP addresses will be dropped immediately and silently.

Trunk security

SIP Trunk Authentication

Register Based Authentication: Many SIP Trunk Service Providers will require a level of Authentication within the SIP Trunk. The Service Provider requires Registration Authentication and Call Initiation Authentication from the PBX. When the PBX initiates a call to the Service Provider, the PBX must provide Authentication within the SIP Protocol for the Service Provider to accept and process the call.

IP Based Authentication: Because some SIP Trunk Service Providers do not support the SIP REGISTER method, you'll need to set up Trunk as the IP Based and add Trunk IP addresses as trusted peers in PBX, then the PBX to accept SIP traffic from trunk IP does not challenge for authentication credentials.

PortSIP PBX supports both Register Based and IP Based Authentication Trunks, but the IP Based Authentication trunk is strongly recommended, it's more secure.

PortSIP PBX is also supporting accepting the Trunk/E1/T1 gateway registration. For example, if an E1/T1 gateway is located in a local LAN but the PBX is in the cloud, we can create an Accept Register" Trunk in PortSIP PBX, set the username and password, and the E1/T1 gateway will be able to use that username and password register to the PortSIP PBX, the PBX only allows make & accept calls with E1/T1 gateway after successfully authorized.

Max concurrent calls limited

PortSIP PBX provides a feature that allows you to set a limit on the maximum number of concurrent calls at both the global and tenant levels for a trunk. If a trunk has already reached its maximum concurrent call limit, any new call attempts will not be processed. This feature ensures efficient call management and prevents overloading of the system.

Outbound route permission

When creating the outbound rule in the PortSIP PBX, you will need to consider outbound rule permission for different users. You can create the outbound rule using the given called number prefix, called number length, and caller belonged user groups.

For example, you can set up outbound rules as below.

• The outbound rule for local calls, long-distance calls, and international calls
  • Create an outbound rule and select the trunk that is least-cost for local calls, and set the user role as Standard International User, then that user will have permission to make calls to the trunk.
• In the menu Blacklist and Codes > Codes and E164, you can find Allowed Country Code and Disallowed Codes options that let you block the calls based on the country code.
• Office hours for the outbound rule
  • PortSIP PBX allows specified office hours for an outbound rule, once set, the outbound rule will be unavailable and no one can make the call on it if outside of those hours.

Network security

Separate Voice Traffic and Data Traffic for some VoIP ISPs, and provide dedicated SIP trunks that support NGN ports (Next Generation Network). NGN can separate data, voice, and video networks or any combination of the three to form a converged network.

For the on-premise deployment, the best practice is to suggest setting up VLAN (Virtual Local Networks) on the PBX. VLAN can improve the call quality but also can secure PBX. The voice traffic and data traffic can be logically separated by a VLAN switch. If one VLAN is penetrated, the other will remain secure. Also, limiting the rate of traffic to IP telephony VLANs can slow down an outside attack.

Transport security

TLS and WSS for SIP signaling

Transport Layer Security (TLS) is a mechanism for securing SIP connections. It is recommended to use TLS as PortSIP PBX SIP transport to prevent data from being passed between other SIP endpoints and PortSIP PBX.

For the WebRTC client, PortSIP offers WSS transport (WebSockets over SSL/TLS). WSS is encrypted, just like HTTPS, and so protects against man-in-the-middle attacks. If the transport is secured, a range of attacks against WebSockets becomes unfeasible.

SRTP and DTLS-SRTP for audio and video

PortSIP PBX and PortSIP Apps support SRTP and DTLS-SRTP. SRTP extends RTP to include encryption and authentication so that all SIP and WebRTC conversations are as secure as possible. The audio and video media data is transported and protected by SRTP/DTLS-SRTP with AES-256 encryption.

Web access security

PortSIP PBX provides HTTPS and HTTP access on the port 8887 and 8888. The following are the recommended practices for securing web portal transactions and preventing unwanted access.

• Create the security rule/firewall rule to disable the HTTP access on TCP port 8888
• Disable Redirect from port 80
• Disable Redirect from port 443
• Upload the trusted SSL certificates, for example, purchase an SSL certificate from DigiCert, GeoTrust

The post Security, Privacy, and Compliance first appeared on PBX & Collaboration & WebRTC & Video Meeting.

Enter the PortSIP PBX SIP Header Manipulation feature, designed to address this challenge seamlessly.

Understanding SIP Header Manipulation

SIP header manipulation empowers administrators to add, delete, or modify SIP message attributes within the PortSIP PBX. These attributes include SIP headers and SIP header elements. But why manipulate them? The primary reasons for doing so are:

1. Compatibility Resolution: Sometimes, two SIP endpoints encounter compatibility issues. By adjusting SIP headers, we can bridge the gap and ensure smooth communication.
2. Special Information Passing: SIP headers allow us to convey specific details during call setup.

How Does PortSIP PBX Handle Tenant Identification?

Let’s delve into the practical application. Imagine an extension within the PBX making a call to a PSTN number. The PBX routes this call to the trunk by sending an INVITE message. Our goal is to easily identify which tenant initiated the call.

Here’s how it works:

1. Adding a Custom SIP Header: PortSIP PBX allows us to insert a custom SIP header, such as “X-Tenant-Name”. We’ll use the tenant’s name as the value for this header.
2. Incorporating Tenant Information: When the PBX sends the INVITE message to the trunk, it includes the “X-Tenant-Name” header with the value set to the tenant’s name. This inclusion ensures that the trunk recognizes the originating tenant.
3. Example Scenario: Suppose the tenant’s name is “ExtremeComm”. The PortSIP PBX will add the “X-Tenant-Name” header with the value “ExtremeComm” to the INVITE message when sending it to the trunk.

Visual Confirmation

In the screenshot above, you can see the settings where the PortSIP PBX adds the “X-Tenant-Name” header with the value “ExtremeComm”. This straightforward configuration ensures that tenant identification becomes effortless.

Beyond Tenant Identification

Remember, the PortSIP PBX SIP header manipulation feature isn’t limited to tenant identification alone. You can also relay, modify, or delete SIP headers based on your specific requirements.

In summary, PortSIP PBX empowers administrators to streamline tenant identification and enhance communication within a multi-tenant environment. Whether you’re fixing compatibility issues or passing special information, SIP header manipulation plays a crucial role.

The post Simplifying Tenant Identification: PortSIP PBX SIP Header Manipulation first appeared on PBX & Collaboration & WebRTC & Video Meeting.

Once someone parks a call at a spot, subscribers who have subscribed to this spot will receive a NOTIFY message for the Dialog Event, and the BLF key will flash. Then the user can press the BLF key to retrieve the call.

What's the Problem of the Normal Call Parking?

As described above, the call parking implementation works fine for traditional scenarios with single-tenant PBX.

However, today is the cloud age, and most scenarios are for the cloud PBX. For multi-tenant usage, a service provider hosts the cloud PBX and creates many tenants. Each tenant has a virtual PBX but shares the same server resources. A cloud PBX instance may handle 1k ~ 10k tenants.

Assume each tenant creates 10 park spots, which means for a cloud PBX with 1k tenants, there will need to be 10k spots created for these tenants. As mentioned above, these spots are actually PBX extensions that need to register to the PBX and maintain registration by sending REGISTER messages periodically. Users need to subscribe to the spots for the Dialog Event, these will consume massive CPU, memory, and bandwidth resources and reduce cloud PBX performance. This is unacceptable for service providers. It’s hard to imagine if tenants need to create more park spots!

Enhanced Call Park Feature in PortSIP PBX

PortSIP PBX is designed for cloud PBX and service provider needs. It provides a real multi-tenant PBX where a service provider just needs to set up one PBX instance, then it has the ability to create thousands of tenants, each tenant is isolated and transparent to each other.

PortSIP implements the call parking feature in a unique way in order to avoid traditional parking problems. It does not require creating park spots and there is no need to subscribe to the Dialog Event package, which we called it “Enhanced Call Park Feature”, it works similarly to the Broadsoft Enhanced Call Park and Webex Call Park Feature.

Here are the details of the PortSIP Call Parking implementation:

• There is no need to create a parking spot. Just use the extension number as the parking spot. For example, if the user wants to park a call to the extension who has extension number 103, just transfer the call to *68103. This *68 is the FAC for Call Park.
• Once a call has parked at an extension, this extension device (IP Phone or softphone app) will receive an out-of-dialog NOTIFY message with the park-info event. The NOTIFY message includes the parker, parked, and retrieve information.
• The extension device can parse parked call details and retrieve information from the NOTIFY message and alert the extension that there is a call parked.
• The user can retrieve the call by pressing the button/soft key.
• There also supports park a call to a park group without the need to specify the parked extension number

Advantages

As listed above, the PortSIP solution has the following advantages:

• No need to create parking spots (extensions), and no need to register massive spots to the PBX and maintain the registration.
• No need to subscribe to the Dialog Event Package for the parking spots, and no need to maintain the subscription.
• The NOTIFY message for the parked call includes detailed information that the IP Phone and softphone app can use to alert the user with park call information and retrieve it with one click.
• For large tenants in the PBX, the call parking feature will not consume massive hardware resources and will not reduce performance.

This article includes the following topics:

• Using Call Parking Feature
• Using Enhanced Call Park on Fanvil IP Phones
• Using Enhanced Call Park on Yealink IP Phones
• Using Enhanced Call Park on Dinstar IP Phones

The post Why Need Enhanced Call Park Feature for Cloud PBX first appeared on PBX & Collaboration & WebRTC & Video Meeting.

This strategic partnership between PortSIP and Fanvil is designed to offer enterprise customers and channel partners a reliable, cost-effective, and cutting-edge Unified Communications system. The integration enables rapid and easily managed deployments, capitalizing on the latest offerings from Fanvil’s SIP devices.

After thorough testing, the following models have been verified for zero-touch auto provisioning with PortSIP PBX. The supported Fanvil devices include:

• X series: X1S/X1SG/X2/X210/X210-V2/X210i-V2/X2C/X3S/X3SG/X3SG Lite/X301(P)(G)(W)/X303(P)(G)(W)/X3SG Pro/X3S(P) Lite/X3S(P) Pro/X3U/X3U Pro/X4/X4U/X4U-V2/X5U/X5U-V2/X6U/X6U-V2/X7/X7-V2/X7A/X7C/X7C-V2
• A series: A320/A320i/A10/A10W
• W series：W611W/W610W/W712
• V series: V62/V63/V64/V65/V67
• H series: H1/H2U-V2/H3/H5/H3W/H5W
• i series: i10S/i10SV/i10SD/i16S/i16SV/i61/i62/i63/i64/i55A/i56A/i57A
• Y Series: Y501(W)/Y501-Y(W)
• FH-S01
• PA Series: PA2S/PA3
• GA series: GA10/GA11

“Fanvil’s products have gained global recognition, and we’re thrilled to have all series integrated with our system,” said Peng Bo, CEO of PortSIP. "Fanvil is a leading brand worldwide, and its devices are as robust as they are elegantly designed. We’re excited about our joint achievements so far, and we eagerly anticipate further growth in our relationship as both companies continue to enhance our business communications offerings.”

"PortSIP PBX is a software-based Multi-tenant PBX that takes full advantage of resources like hardware, manpower, etc. to offer customers unmatched Unified Communications," said Louis Chen, Executive President of Fanvil. "Thanks to the active and close cooperation, we have finished the compatibility test for all of our SIP devices, which significantly smoothens the operating experience for both PortSIP and Fanvil users. We are deeply grateful for PortSIP, whose professionalism, efficiency, and responsibility are noticeable readily. Cheers for the remarkable achievement and the long-term close partnership."

About PortSIP (www.portsip.com)

PortSIP delivers all-in-one modern unified communications solutions to service providers, enterprises, and critical infrastructure sectors globally. Clients of PortSIP include HPE, Qualcomm, Agilent, Keysight, CHUBB, Netflix, Nextiva, FPT, Panasonic, Softbank, Telstra, T-Mobile, Siemens, BASF, Queensland Rail, and others. We engage deeply with our customers, helping them modernize their communications for improved competitive positioning and business outcomes in today’s smart, always-on, and data-hungry world.

About Fanvil (www.fanvil.com)

Fanvil is a leading global provider of Audio&Video-IoT (A&V-IoT) devices. With three R&D centers in Beijing, Shenzhen, and Suzhou, China, Fanvil has compiled an effective team of R&D, production, sales, and service staff to innovate and add value to our business partners. As the pioneer in applying standardized network communication technology and audio and video technologies to build A&V-IoT, Fanvil is boosting the digital transformation for multiple industries.

The post PortSIP and Fanvil Announce Certified Interoperability of Fanvil’s IP Phones and Intercom Devices with PortSIP PBX first appeared on PBX & Collaboration & WebRTC & Video Meeting.

Troubleshooting SIP messages and VoIP call quality can be tough. Call drops delays and unexpected silences can harm your credibility and result in lost business opportunities.

PortSIP Trace Server is designed from scratch to offer all the essential features for modern Telco. It comes packed with advanced features like Real-Time Session Tracking, RTP Analyzers, Partitioning, Clustering, and more. It's fully compatible with our existing Open-Source HEP Stack and Agents, offering the best of both worlds.

Key Benefits

• Access singularly to retrieve SIP and RTCP QoS captures via Web UI
• Centrally store SIP capture data across many hosts
• More intuitively filter SIP capture data and correlate the data to the dialog/transactions each request/response is part of (this is immensely useful!)
• Gracefully age dated capture data you don’t want to persist for very long (though persisting longer term can be configured very easily)
• The calls and RTCP data are beautifully displayed as charts.

Troubleshoot the call by observing the call flow

Checking Call Information

Perform Quality of Service (QoS) analytics 

For more information and guidance on setting up the PortSIP Trace Server, please refer to the article: Trace Server - A Better Way for Monitoring SIP Messages and QoS in PortSIP PBX. It provides comprehensive instructions and details to help you get started.

The post A Better Way for Monitoring SIP Messages and QoS in PortSIP PBX first appeared on PBX & Collaboration & WebRTC & Video Meeting.

The PortSIP PBX v16.2.0 primarily addresses new features and is recommended for all users.

Key Updates in PortSIP PBX Release v16.2.0:

• Supports SNOM M series and Yealink W series DECT phones.
• Adds support auto provisioning for Fanvil i504(W), i505(W), i506W, and i507W devices.
• Changes the SNOM phone configuration file to HTM format.
• Enables playing of call recording voice prompts for inbound calls.
• Allows setting of office hours and holidays for each IVR DTMF input.
• If an extension is registered to PBX from multiple devices, once one of the devices rejects the incoming call by 486, the PBX will also hang up the call on other ringing devices.
• Adds an option for the trunk to allow or disallow the PBX to adjust the SDP direction when holding the call.
• Supports auto-provisioning of emergency numbers to the IP phones, allowing the IP phone to dial the emergency number even when the phone keys are locked.
• No longer displays the offline agent in the Contact Center Wallboard.
• Removes the Offline status in the BLF settings.
• Supports creating the transport on the port which is less than 1024 (limited permissions by some Linux).
• Displays the Feature Access Code in the WebRTC and Windows Client.
• Updates the apps (iOS, Android, Windows, WebRTC) to synchronize the status between the apps and IP Phones. The statuses include:
  • Online
  • Ringing
  • On call
  • Away
  • Do Not Disturb
  • Business Trip
• Supports synchronization of the Do Not Disturb (DND) status by pressing the DND button on the IP phone.
• Adjust the REST API rate limit to 5,000 per minute.
• Changed the WSI (Web Socket Interface) events so subscribers more easily watch the extension status.
•  Fixes the bug that caused the SNOM phone not to alert for new voicemail.

Users running PortSIP PBX v16.0.x or v16.1.x are advised to follow the provided guide for upgrading the PBX: Upgrading PortSIP PBX v16.x to the Latest Version.

PortSIP SBC v10.0.9

In addition to the PBX release, PortSIP has also rolled out the SBC v10.0.9. This minor release is focused on enhancing the user experience for WebRTC clients.

The following changes are included in this release:

• Updated the WebRTC client App to support synchronous the extension status with other Apps and IP Phones
• Improve performance
• Fixed minor bugs

Users are advised to follow the guide titled Upgrading PortSIP SBC v10.x to the Latest Version for a smooth upgrade process.

The post PortSIP PBX v16.2.0 is Ready first appeared on PBX & Collaboration & WebRTC & Video Meeting.

• They require additional engineers to maintain a large number of PBX instances.
• The pseudo-multi-tenant architecture results in substantial wastage of hardware resources.

Consider a service provider catering to 1,000 clients. With a pseudo-multi-tenant PBX, the provider would need to establish a new PBX instance for each client (tenant). Assuming each tenant has approximately 10–50 extensions, or even 100 extensions, the hardware resources and maintenance efforts required are staggering!

PortSIP PBX, on the other hand, offers a genuine multi-tenant PBX solution. Unlike other PBX systems that necessitate a separate PBX instance for each tenant, PortSIP PBX enables service providers to establish a single instance and create over 100K tenants within that PBX instance. Each tenant can have its own settings and extension numbers, remaining invisible to other tenants. It gives the tenant the impression of having an independent PBX. This feature is a significant advantage as it conserves hardware resources and reduces maintenance efforts.

One-Click Tenant Management

PortSIP PBX offers a streamlined tenant management system. All tenants are conveniently listed in the "Tenants" menu, providing system administrators with easy access to manage each tenant. Administrators can effortlessly enable or disable a tenant, or modify a tenant’s capabilities. These capabilities include:

• Storage size
• Enable/disable call recording
• Maximum number of extensions, simultaneous calls, ring groups, queues, meetings, and more

This user-friendly interface simplifies the task of managing multiple tenants, making PortSIP PBX an efficient choice for service providers.

Switching Tenant Administrator Role for PBX Management

PortSIP PBX offers a unique feature that allows system administrators (service providers) to switch into a tenant’s scope, assuming the role of a tenant administrator. This enables them to manage the PBX for the tenant, check the tenant’s settings, and then switch back to the system administrator level. This feature significantly reduces the workload associated with PBX management.

As demonstrated in the screenshot, when the System Administrator is in the tenant scope, it’s straightforward to switch back to the system level.

Additionally, a user (extension) within a tenant, who has been assigned the tenant admin role, can also log into the PortSIP PBX web portal to manage their own virtual PBX. However, they won’t have visibility of other tenants. This ensures each tenant maintains its privacy and independence while benefiting from efficient management.

Indeed, a true multi-tenant PBX like PortSIP PBX is designed for the cloud age. It efficiently utilizes hardware resources and simplifies maintenance efforts, making it an ideal solution for service providers in today’s cloud-centric world. This innovative approach aligns with the evolving needs of businesses, offering scalability and flexibility while ensuring each tenant enjoys the benefits of an independent PBX.

To experience the genuine multi-tenant functionality of PortSIP PBX, designed for the modern cloud communication era, you can download and install the fully-featured free version of PortSIP PBX. This will allow you to explore its robust features and understand how it can streamline your communication needs. Enjoy exploring!

PortSIP PBX Solution Has Your Back

When weighing the solution for hosting the Cloud PBX, you have to look at your needs today and tomorrow. You have to ask yourself where you want to be in the next year or two and beyond.

Here at PortSIP, we’ll walk you through everything, including free porting of your existing solution. We’ll help you select the right VoIP desk phones that meet your client's budget and everyday business needs.

In the age of the cloud, you cannot keep inching along with a legacy PBX. Your traditional PBX solution costs you more than it’s saving.

Take advantage of the PortSIP Solution and the freedom to work from anywhere. PortSIP PBX lets you focus on serving your customers, not trying to figure out the system. You can leave the rest to us.

The post Multi-tenant PBX: Genuine or Pseudo? first appeared on PBX & Collaboration & WebRTC & Video Meeting.

For service providers, a cloud PBX must support a large number of users and handle high volumes of calls. Typically, a cloud PBX should be based on a multi-tenant architecture, allowing the service provider to set up a single PBX installation that can serve a large number of tenants, users (extensions), and calls. Therefore, performance and scalability are crucial for a cloud PBX solution.

Most PBXs in the market address these concerns in the following ways:

1. Deploy multiple PBX instances and place SIP Proxy servers (such as OpenSIPS, and Kamailio) in front of the PBXs. These are usually open-source PBX solutions like Asterisk, FreeSwitch, or PBXs (for example, Bicom) based on Asterisk/FreeSwitch. The downside is that the configuration is complex, requires senior engineers for maintenance, and consumes substantial hardware resources.

2. Deploy multiple PBX instances, set each PBX to other PBXs as the trunk to build a mesh network, and then assign users to different PBX instances. This solution is difficult to manage and also consumes substantial hardware resources.

PortSIP Solution

PortSIP has developed its PBX and UCaaS in-house, focusing on high performance and designed specifically for the cloud communication age. The architecture is 1-N, with the central server handling only the SIP signaling. Other application servers such as Media Server, Queue Server, IVR Server, and Meeting Server can be deployed separately on other servers. These server components support scalability. With the following architecture topology, a PortSIP PBX installation can support a maximum of unlimited tenants, 100K users, and 10K simultaneous calls.

The PBX administrator simply needs to launch an application server instance, and then add the server information in the PBX web portal. The PBX load balancer will then allocate the meetings, calls, and RTP streams to different servers. This can be done with just a few clicks, without needing extensive technical knowledge or skills.

PortSIP UCaaS

For those with more than 100K users, you may want to check out the PortSIP UCaaS solution. It's designed for large service providers, supporting 1M+ users by deploying Kubernetes. It's easy to scale to support more users by simply adding more servers. To satisfy all the Unified Communications-critical features, it provides enterprise users and service provider customers with a powerful way to grow revenue and tap new markets.

PortSIP PBX Solution Has Your Back

When weighing the solution for hosting the Cloud PBX, you have to look at your needs today and tomorrow. You have to ask yourself where you want to be in the next year or two and beyond.

Here at PortSIP, we’ll walk you through everything, including free porting of your existing solution. We’ll help you select the right VoIP desk phones that meet your client's budget and everyday business needs.

In the age of the cloud, you cannot keep inching along with a legacy PBX. Your traditional PBX solution costs you more than it’s saving.

Take advantage of the PortSIP Solution and the freedom to work from anywhere. PortSIP PBX lets you focus on serving your customers, not trying to figure out the system. You can leave the rest to us.

The post How is PortSIP PBX Dealing with Large Volume Calls for Service Providers first appeared on PBX & Collaboration & WebRTC & Video Meeting.

This minor release, PBX v16.1.2, primarily addresses bug fixes and is recommended for all users.

Key Updates in PortSIP PBX Release v16.1.2:

• An option has been added to use the last six digits of the IP Phone’s MAC address as the IP Phone web UI password.
• A bug has been fixed that affected the search functionality for contacts by name, display name, and extension number.
• A bug has been fixed that affected the synchronization of instant messages between apps when a user was logged into the mobile app, WebRTC client, and Windows app simultaneously.
• A bug has been fixed that prevented the G.729 codec from being enabled for the provision of SNOM phones.
• A bug has been fixed that caused the BLF key to not work with Fanvil W611W.
• A bug has been fixed that caused one-way audio with some trunks if the PBX server OS had IPv6 disabled.
• The voice prompt for announcing the queue position in Italian has been improved.

Users running PortSIP PBX v16.0.x and v16.1.x are advised to follow the provided guide for upgrading the PBX: Upgrading PortSIP PBX v16.x to the Latest Version.

PortSIP SBC

In addition to the PBX release, PortSIP has also rolled out the SBC v10.0.8. This minor release is focused on enhancing the user experience for WebRTC clients. Users are advised to follow the guide titled Upgrading PortSIP SBC v10.x to the Latest Version for a smooth upgrade process.

The post PortSIP PBX v16.1.2 is Ready first appeared on PBX & Collaboration & WebRTC & Video Meeting.

Between PortSIP PBX vs other PBXs for a business unified communications solution, what would you choose?

Real Multi-Tenant PBX Designed for Cloud Age

Many vendors claim to provide multi-tenant PBXs, but most of their architecture is pseudo-multi-tenant. Typically, they need to install a new PBX instance for each tenant and then manage these instances through a centralized web portal. End-users are unaware of this, but for service providers, this is a significant issue:

• They need to maintain massive PBX instances.
• They waste server hardware resources.

For example, if a service provider has 1000 clients, each client is a tenant and needs to set up a PBX instance. Assume each tenant has about 10–50 extensions, and then the server resources and maintenance work are unimaginable!

PortSIP PBX provides a real multi-tenant PBX. Unlike other PBXs that require a separate PBX instance for each tenant, PortSIP PBX allows service providers to set up one instance and create thousands of tenants in that PBX instance. Each tenant can have its settings and extension numbers and is invisible to other tenants, the tenant just feels he has a whole PBX. This is an excellent feature that can save hardware resources and reduce maintenance work.

As shown in the screenshot above, the system administrator (service provider) or dealers (distributor, sub-distributor, reseller) can view and manage all tenants in the web portal. The following actions are available:

• Manage. Switch to manage the tenant's extensions, call routings, settings, etc.
• Add. Create a new tenant.
• Edit. Change the settings of the tenant. For example, limit the capability of that tenant.
• Disable. Disable the tenant temporarily. All settings and data will be preserved and can be enabled again at any time.
• Delete. Delete the tenant from the PBX. All data of that tenant will be erased and can’t be recovered.

This is what modern multi-tenant PBX architecture is designed for in the cloud age. PortSIP has done an excellent job with this, and our clients are enjoying it a lot.

Full White-label Solution

Building your brand is never easy, but PortSIP can help you get there. Our rebranding feature allows you to fully private label your PBX and cloud PBX service and make it your own, including the following elements:

• Theme
• Product Name
• Company Name
• Website Link
• Logo Favicon
• User Agent string of PortSIP PBX and SBC
• WebRTC, Windows, and Mobile Apps
• IP Phone Templates
• Notification Email Templates

You just need to focus on the marketing, sales, and billing of your services, and we'll take care of the rest! Please refer to the article Rebranding PortSIP PBX, SBC for more details.

A Scalable Communications Platform

For cloud multi-tenant PBXs, the service provider hosts the PBX in the cloud and provides the calling service to users. Unlike single-tenant PBXs for SMBs, this means massive tenants and simultaneous calls are handled by the cloud PBX. The PBX vendor must take care of the performance.

PortSIP PBX is designed for the cloud age and offers high performance. It can handle massive extensions (users) and simultaneous calls. For large-size deployments, the PBX can extend the below servers for the cluster:

• Queue Server
• Meeting Server
• IVR Server
• Media server

With the extended servers cluster, the PBX Call Manager server only needs to process signaling. Utilizing multiple threads and cache technologies enables a PortSIP PBX instance to handle up to 100K users and 20K simultaneous calls. This is an impressive feat that meets most service providers’ scenarios.

As shown in the screenshot above, we disabled the default media server in the PBX server and extended three media servers in order to handle heavy calls.

Designed for High Performance

When designing a cloud PBX, every feature needs to be considered deeply, since it may cause performance issues when the amount of tenants and users is large.

For instance, for the call park feature, as per traditional PBX design, the PBX needs to create a dozen park spots for each tenant. In reality, these spots are PBX extensions that need to register with the PBX and refresh the registration period. The extensions also need to subscribe to these spot extensions for the dialog event in order to receive an alert when a call has been parked on the spot. Additionally, the extensions need to maintain their subscription by sending the SUBSCRIBE message periodically.

Assume each tenant creates 5 park spots, then for a cloud PBX with 1k tenants, there will need to be 5K park spots (extensions) created for the tenants. As mentioned above, these spots are actually PBX extensions. The registration and subscription of these spots will consume massive CPU, memory, and bandwidth resources and reduce cloud PBX performance. This is unacceptable for service providers. It’s hard to imagine if tenants need to create more park spots!

PortSIP pays attention to each feature for multi-tenancy. In order to avoid the problems of traditional PBXs, PortSIP has designed the call parking feature in a unique way that is easy to use and more friendly. It also has good performance and uses fewer resources. You can find more details in these articles:

• Using Enhanced Call Park on Fanvil IP Phones
• Using Enhanced Call Park on Yealink IP Phones
• Using Enhanced Call Park on Dinstar IP Phones

Seamless Migration from BroadSoft

PortSIP PBX also offers rich features that are different from other PBX, a lot of the features are like the Broadsoft, for example, the Feature Access Codes (dial codes).

These PortSIP PBX feature access codes are mostly the same as Broadsoft and allow users to work more easily. Each tenant can customize its own feature access code to meet their user experience needs. You can find more details in this article: Feature Access Codes.

Modern Unified Collaboration Solution

PortSIP PBX also offers a free comprehensive collaboration and communication solution for hybrid work, enabling inclusive experiences for everyone while protecting your data with industry-leading security. It provides calling, messaging, meetings, and sharing in a unified experience across web, desktop, and mobile apps. All the ways to connect:

• Audio and Video Calling
• Audio and Video Meetings
• Messaging and Presence
• Audio Messaging
• Video Messaging
• Screen Sharing
• File Sharing

Effortless User Management

The tenant administrator can manage users easily through the user list in the PBX web portal, where all extension statuses are listed and indicate whether they are online/offline/on call/push notification enabled/automatic callback enabled/DND enabled.

You can click on the search icon to check an online extension's device details.

As shown in the below screenshot, extension 102 is registered from a PortSIP UC client (IP is 192.168.0.22, the port is 5960 over the UDP transport), and the Yealink T53 IP Phone (IP is 192.168.0.36, the port is 5060 over the UDP transport).

CDR and Recording

PortSIP PBX supports both audio and video recording and can record calls in MP3 or MP4 format, which makes it easy to play them in the browser.

The CDR is linked with the recording file, which can be downloaded and played in a browser.

The PortSIP CDR feature is different from other PBXs in that if a call is rerouted and forwarded multiple times during the call, the PBX will tie all call targets in one CDR. This makes it easy to track the call flow.

For example, if a call comes from the SIP trunk and arrives at the PBX, then is routed to the IVR and re-routed to the queue by the user pressing DTMF, and an agent of the queue answers that call, all this information(callee number, callee IP address, and port) and timestamp are recorded in the CDR. For more details, please read this article: CDR and Call Recording.

Privacy

PortSIP PBX also takes care of the privacy of the recording. Due to privacy and security concerns in some countries, the law stipulates that when a call is made between two external numbers, it should not be recorded.

Consider the following scenario: The client calls the contact center from the trunk and the agent answers. The call is starting to be recorded, and after a while of conversation, the agent transfers the client’s call to another landline/mobile phone number, the call is now between two PSTN phone numbers. At this point, the PBX should stop the call recording automatically. PortSIP provides an option for this feature, please read the details in this article: Automatically stop recording the call between two external numbers.

Security

Essentially, the cloud PBX phone system is a communication solution that enables your business to make and receive calls via the Internet, thereby rendering traditional landlines obsolete. However, like any cloud-based system, it introduces new security vulnerabilities, such as cyber-attacks and phishing scams.

PortSIP PBX places a high priority on security to keep your business safe. For instance, in PortSIP PBX, each extension’s IP Phone configuration file is stored in a distinct directory with a randomly generated long name. This prevents anyone from guessing the URL to download the configuration file, even if the phone’s MAC address is leaked. This approach sets PortSIP PBX apart from other PBX systems.

Please read the article PortSIP PBX Security Features for more details.

Trunk Management

Different cloud PBX providers have varying requirements due to their unique business models.

Some providers only host the cloud PBX and do not offer SIP trunks to tenants, requiring tenants to configure their own trunks.

Other service providers offer cloud PBX and SIP trunk bundles, limiting tenant configuration of trunks. In this case, all trunks must be configured by the service provider, who assigns a range of DID numbers (DID pool) to tenants. Tenants can only use the trunk and DID numbers assigned by the service provider to create call routing rules.

Some service providers offer a hybrid mode that allows tenants to configure their own trunks and use the trunks and DID numbers assigned by the service provider.

PortSIP PBX supports all requirements for service providers to meet their business models. An option called Enable Tenant Level Trunk allows service providers to enable or disable tenant trunk configuration.

Pub/Sub

PortSIP PBX provides the Pub/Sub mechanism which is based on the WebSocket (PortSIP WSI).

The user is able to create the WebSocket in any programming language to subscribe to the PBX events, once the subscribed events occur, PortSIP PBX will send the event message to the subscriber automatically, the message is in the JSON format. For more details please read this article: Going Real-Time with PortSIP PBX Pub/Sub.

Dealers

The cloud PBX/UC service providers just need to focus on hosting the PBX system in the cloud, ensuring service operation, and providing continuous stable service.

Distributors/Resellers partner with the service providers to give out cloud PBX services and products to the customers.

A distributor or the reseller purchases cloud PBX services and cloud-based PBX products from the cloud service provider and further sells them to the end users.

PortSIP PBX offers a great feature called Dealers for the above scenario that allows service providers to manage dealers (distributors and resellers) easily. For more details please read the article: Dealers.

Free SBC

PortSIP PBX offers a free SBC for secure deployment and also offers WebRTC and Microsoft Teams Direct Routing. The SBC can be deployed as a cluster to support large call volumes, those are totally free.

Microsoft 365 Integration

PortSIP PBX seamlessly integrates business communications into your cloud work environment, transforming Microsoft 365 into an enterprise-grade communications hub with a powerful and easy-to-use interface and SSO, that is simple for end-users to deploy and IT to manage.
For more details, please read this article: Microsoft 365 Integration.

Push Notifications

A Voice over Internet Protocol (VoIP) app lets users make and receive phone calls by using an Internet connection instead of the device’s cellular service. Since the VoIP app relies heavily on the network, it’s not surprising that making calls usually results in high power consumption.

PortSIP PBX supports mobile push notifications for iOS and Android apps. All PortSIP users will benefit from this feature, for more details please read this article: How Do Push Notifications Work with PortSIP PBX?

Free PortSIP VoIP SDK

PortSIP offers a VoIP SDK that can be used with PortSIP PBX to create modern UC apps for iOS, Android, Windows, macOS, and WebRTC.

This award-winning SDK is used by many large companies such as T-Mobile, Nextiva, Qualcomm, Telstra, HPE, Siemens, Netflix, Agilent, and Dialogic: PortSIP VoIP SDK.

Rebranding App

PortSIP PBX has an app for the below OS:

• iOS
• Android
• Windows Desktop
• WebRTC

Open REST API

PortSIP PBX provides full REST API which means customers can easily integrate it with the current business. In fact, the PortSIP PBX Web Portal is also built based on this REST API

PortSIP PBX Solution Has Your Back

When weighing the solution for hosting the Cloud PBX, you have to look at your needs today and tomorrow. You have to ask yourself where you want to be in the next year or two and beyond.

Here at PortSIP, we’ll walk you through everything, including free porting of your existing solution. We’ll help you select the right VoIP desk phones that meet your client's budget and everyday business needs.

In the age of the cloud, you cannot keep inching along with a legacy PBX. Your traditional PBX solution costs you more than it’s saving.

Take advantage of the PortSIP Solution and the freedom to work from anywhere. PortSIP PBX lets you focus on serving your customers, not trying to figure out the system. You can leave the rest to us.

The post The Advantages of PortSIP PBX vs. Other PBXs first appeared on PBX & Collaboration & WebRTC & Video Meeting.