User Interface (Without High Availability Mode)

Updated at May 13th, 2024

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Table of Contents

Overview Configuration Networking Info Basic SIP Configuration Cipher Suite Name supported for TLS Transport Firewall Configuration NAT Configuration Not Supported Features

Overview

The Overview tab gives the basic information of the current User’s settings as well as their current status.

Item	Description
Label	For mnemonic, customers can modify the label of their SIP username
Subscription UUID	User can click on UUID to copy to locate multiple SIP Trunk licenses in the Licenses tab under Manage Organization
Type	Normal: when the selected SIP trunk is normal Carrier Connect: when the selected SIP trunk is with Carrier Connect
Authentication Mode	Method customer using for authentication, we currently support 2 types Account (Digest Authentication): Default mode IP Authentication: Available when having IP Authentication Mode add-on
Server Location	Server location is dependent on the country
SIP Server	Shows full SIP domain on SIP Server field
CA Certificate	This certification will be used to configure the client side in case the client side needs to trust the server manually
Members	Members granted access to configure the SIP account
Assigned Numbers	DIDs number purchased from B3 partner

Configuration

Authentication Mode

Account (Digest Authentication): Show SIP credentials information (Username and Password)

IP (IP Authentication): IP configuration is required to log in SIP account to device

Reset SIP Password

Background:
Once users click on the Reset button, they can either input a password manually or generate a random password by clicking on the generate icon.

Password Conditions:

The following conditions should be fulfilled for correct passwords:

Password must have 14 characters
Password must have at least one uppercase letter
Password must have at least one number
The copy icon is enabled when users enter a correct password, fulfilling the above criteria
Users can not input any characters once the character limit is reached (Maximum 14 characters)

Security

User-interface for both Primary Sip Account and Secondary Sip Account will depend on the Authentication Mode selected for a particular user/SIP account.

If the Authentication Mode is Account, UI will be displayed as below:

If the Authentication Mode is IP, then the Security section will not be applicable and the UI will be displayed as below:

Call Rules

Inbound Call Rule: This function will help the customer customize their own incoming call display Caller ID

Outbound Call Rule: This function will help the customer customize their own outgoing call dial plan

Main Trunk Number

Outgoing calls will use the user's sent Caller ID if it matches one on the purchased list.
Main Trunk Number is only used if the user's sent Caller ID is not on their purchased number list. In this case, the Caller ID for outgoing calls is the selected Trunk number.
If SIP Trunk account is assigned multiple numbers, the system will detect the 1st assigned number to be the Main Trunk Number

Default Caller ID

Default Caller ID has two categories to use as Caller ID:

SIP trunk Numbers
- Private
- Number(s) assigned to this SIP Trunk ( for instance, +6566xxxxx…etc)

Auto Attendant Numbers
- Number(s) assigned to Auto Attendant

SIP Trunk Caller ID Logic

The system will assess whether the PBX sent caller number is VALID or not. If it's valid ( within the range of DIDs assigned to that SIP trunk), the system will prioritize using the PBX caller number as the outgoing caller ID.
If the number is INVALID (not within the range of DIDs assigned to the SIP trunk), the system will prioritize using the current Main Trunk Number setting as the caller ID.

Examples:

Valid CallerID send from PBX use case

The SIP trunk has two numbers assigned, 65123456 and 65999999, with the Main Trunk Number set to 65123456 / private and the PBX sends 65999999 as the caller ID, the system recognizes 65999999 as valid and continues to use 65999999 as the caller ID, regardless of the 65123456 / private Main Trunk Number setting.

Invalid CallerID send from PBX use case

If the SIP trunk has two numbers, 65123456 and 65999999 with the Main Trunk Number set to 'private' and the PBX sends 1234567 as the caller ID, the system recognizes 1234567 invalid. In this case, it will set the caller ID based on the current Main Trunk Number setting and send 'private' as the caller ID.

Manual Fail-over

When this mode is activated, it allows users to forward all the incoming calls of the SIP number(s) to the backup number(s). All the calls will be forwarded to the backup number(s) without checking the SIP account status.

Master Backup Line Number: An external number to receive calls when the SIP phone lines are down. All incoming calls to any SIP numbers that do not have a backup number will go to the Master Backup Line Number.

Backup Number Customization: Users can customize to forward incoming calls of each SIP Number to a particular Backup Number.

Networking Info

This section shows parameters related to network requirements for the customer side including firewall configuration, NAT configuration and the features which are not supported by the SIP account.

Basic SIP Configuration

Item	Description
Codecs	Our SIP trunk supports two audio codecs – G711A on port 5061 and Opus on port 5081. Codecs determine how voice communication is encoded and transmitted.
Dial Pattern	We accommodate the dialed format of Country Code + Area Code + Phone Number (e.g., 8498765432, 16505300061) to facilitate easy and standardized calling.
Transport Layer	Our SIP trunk employs the TLS (Transport Layer Security) protocol to ensure secure data transmission over the network. TLS encrypts the communication between devices.
SSL/TLS Protocol Version	Our system supports TLS version 1.2 or newer, ensuring the highest level of security for your data exchange.
SIP Registration Period	We offer a SIP registration period of 3600 seconds, allowing your devices to maintain a consistent connection.
Audio Format	Our SIP trunk supports Secure Real-time Transport Protocol (SRTP) for audio transmission, ensuring the confidentiality and integrity of voice data.
DTMF Mode	Our system supports DTMF (Dual-Tone Multi-Frequency) signaling using the RFC2833 standard, allowing the transmission of touch-tone signals during a call.
SIP Option	SIP options are provided in Basic SIP Configuration to enhance communication interoperability and ensures seamless integration with various SIP-enabled devices and services, optimizing overall system performance and functionality.

Cipher Suite Name supported for TLS Transport

Cipher suite name supported for TLS Transport are provided to ensure that only secure encryption protocols are utilized, mitigating the risk of potential hacking attempts. Without this information, PBX customers might inadvertently select weaker Cipher suites, leaving their system vulnerable to security breaches.

Firewall Configuration

Item	Description
Signalling (SIP)
IP Addresses	Our SIP server's IP address is 52.77.0.62, and it exclusively supports TLS encryption, guaranteeing secure data transmission.
Ports	We utilize port 5061 (TLS with PCMA codec) and port 5081 (TLS with Opus codec) for signalling purposes, ensuring reliable and encrypted communication channels.
Direction	Both incoming and outgoing traffic are facilitated through these ports.
Audio (RTP)
IP Adresses	The IP addresses for audio transmission range from 54.251.255.196 to 54.251.255.211.
Ports	The audio transmission ports span from 10000 to 30000.
Direction	Both incoming and outgoing audio traffic utilize these specified IP addresses and ports.

NAT Configuration

NAT Traversal: We offer support for STUN (Session Traversal Utilities for NAT) to effectively navigate NAT barriers.
STUN Server: For seamless NAT traversal, our recommended STUN server is stun.b3networks.com.

Not Supported Features

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