SIP Call Configuration
Session Initiation Protocol(SIP) is a signaling transmission protocol used for initiating, maintaining, and terminating calls.
A SIP call uses SIP to send and receive data between SIP devices, and can use the internet or a local network to offer high-quality and secure communication. Initiating a SIP call requires a SIP account, a SIP address for each device, and configuring SIP settings on the devices.
SIP Account Registration
Each device needs a SIP account to make and receive SIP calls.
Akuvox intercom devices support the configuration of two SIP accounts, which can be registered under two independent servers.
Click here to view the SIP account registration example.
Register SIP accounts on the web Account > Basic > SIP Account interface.
Account 1/Account 2: the door phone supports 2 SIP accounts.
- Account 1 is the default account for call processing. Also, it will be utilized when the Akuvox SmartPlus cloud service is activated.
- The system switches to Account 2 if Account 1 is not registered.
Tip
When the device is connected to the SmartPlus Cloud, the display label, register name, and username will show its SIP number.
Display Label: The label of the device.
Display Name: The display name of the device.
Register Name: Same as the username from the PBX server.
Username: Same as the username from the PBX server for authentication.
Password: Same as the password from the PBX server for authentication.
SIP Server Configuration
SIP servers enable devices to establish and manage call sessions with other intercom devices using the SIP protocol. They can be third-party servers or built-in PBX in Akuvox indoor monitor.
Navigate to the web Account > Basic interface.
SIP Server Address: Enter the server’s IP address or its domain name.
SIP Server Port: Specify the SIP server port for data transmission.
Registration Period: Define the time limit for SIP account registration. Automatic re-registration will initiate if the account registration fails within this specified period.
Outbound Proxy Server
An outbound proxy server is used to receive all initiating request messages and route them to the designated SIP server in order to establish a call session via port-based data transmission.
Set it up on the web Account > Basic > Outbound Proxy Server interface.
Preferred Outbound Proxy Server: Enter the SIP proxy IP address.
Preferred Outbound Proxy Server Port: Set the port for establishing a call session via the outbound proxy server.
Alternative Outbound Proxy Server: Enter the SIP proxy IP address to be used when the main proxy malfunctions.
Alternative Outbound Proxy Server Port: Set the proxy port for establishing a call session via the backup outbound proxy server.
Data Transmission Type
SIP messages can be transmitted using three data transmission protocols: UDP (User Datagram Protocol), TCP (Transmission Control Protocol), and TLS (Transport Layer Security). In the meantime, you can also identify the server from which the data comes.
Set it up on the Account > Basic > Transport Type interface.
UDP: An unreliable but very efficient transport layer protocol. It is the default transport protocol.
TCP: A less efficient but reliable transport layer protocol.
TLS: An encrypted and secured transport layer protocol. Select this option if you wish to encrypt the SIP messages for enhanced security or if the other party’s server uses TLS. To use it, you need to upload certificates for authentication.
SIP Hacking Protection
Internet phone eavesdropping is a network attack that allows unauthorized parties to intercept and access the content of the communication sessions between intercom users. This can expose sensitive and confidential information to the attackers. SIP hacking protection is a technique that secures SIP calls from being compromised on the Internet.
Set it up on the Account > Advanced > Call interface.
Prevent SIP Hacking: Activate this feature to only receive calls from contacts in the whitelist. This protects users’ private and secret information from potential hackers during SIP calls.
Audio Codec Configuration
The device supports four types of codecs (PCMU, PCMA, G729, and G722) for encoding and decoding the audio data during the call session. Each codec varies in terms of sound quality. Higher bandwidth means the device can capture more detail, leading to clearer sound, and higher sample rates capture more data, reducing distortion and preserving sound quality.
You can select the specific codec with different bandwidths and sample rates flexibly according to the actual network environment.
To set it up, go to the web Account > Advanced interface.
Please refer to the bandwidth consumption and sample rate for the codec types below:
Codec Type | Bandwidth Consumption | Sample Rate |
|---|---|---|
PCMA | 64 kbit/s | 8kHZ |
PCMU | 64 kbit/s | 8kHZ |
G729 | 8 kbit/s | 8kHZ |
G722 | 64 kbit/s | 16kHZ |