Overview
The SIP Settings page lets you configure the core SIP protocol parameters of your Cloud PBX. These settings control how the system handles SIP signaling, audio and video codecs, encryption, session management, Quality of Service (QoS), and fax transmission.
To access these settings, log in to the PBX admin portal and navigate to PBX Settings > SIP Settings.
The page is divided into seven tabs: General, Codec, TLS, Session, QoS, T.38, and Advanced.
Important: Changes to SIP settings affect the entire PBX system. Incorrect values may cause registration failures or call quality issues. Only modify these settings if you understand their impact, or contact Virtual-Call support for assistance.
General
The General tab defines the basic SIP transport ports and registration timers for endpoints and trunks.
Basic
| Setting | Description |
|---|---|
| SIP UDP Port | The port used for SIP signaling over UDP. Default: 5060. |
| SIP TCP Port | The port used for SIP signaling over TCP. Default: 5060. |
SIP Endpoint Registration Timer
Controls how frequently registered endpoints (IP phones, softphones) must re-register with the PBX to remain active.
| Setting | Description |
|---|---|
| Max Registration Time (s) | Maximum interval between registration renewals. Default: 1800 seconds (30 minutes). |
| Min Registration Time (s) | Minimum interval between registration renewals. Default: 600 seconds (10 minutes). |
| Qualify Frequency (s) | How often the PBX sends OPTIONS requests to check if an endpoint is still reachable. Default: 60 seconds. |
Outbound SIP Registration Timer
Controls the registration behavior for outbound trunk connections.
| Setting | Description |
|---|---|
| Registration Attempts | Number of registration retry attempts. 0 means unlimited retries. |
| Default Registration Time (s) | Default interval for outbound trunk re-registration. Default: 1800 seconds. |
SIP Endpoint Subscription Timer
Controls SIP event subscriptions (e.g., BLF status monitoring, presence updates).
| Setting | Description |
|---|---|
| Max Subscription Time (s) | Maximum interval between subscription renewals. Default: 3600 seconds (60 minutes). |
| Min Subscription Time (s) | Minimum interval between subscription renewals. Default: 600 seconds (10 minutes). |
Codec
The Codec tab defines which audio and video codecs the PBX uses and their priority. Codecs determine how voice and video are compressed during calls, affecting both call quality and bandwidth usage.
| Setting | Description |
|---|---|
| iLBC Mode | Frame length for the iLBC codec. Options: 20 ms or 30 ms. A shorter frame length provides slightly better quality at the cost of higher bandwidth. |
| Available Codecs | Codecs that can be added to the selected list. Move codecs to the right panel to activate them. |
| Selected Codecs | Active codecs are listed in priority order (top = highest priority). The PBX negotiates codecs with the remote party in this order. |
Tip: For best voice quality on modern networks, keep u-law (G.711u) and a-law (G.711a) at the top of the selected codec list. These uncompressed codecs provide excellent audio quality and are widely compatible. Add Opus if your endpoints support it—it offers high quality at lower bandwidth.
TLS
The TLS tab configures encrypted SIP signaling. When enabled, SIP messages between the PBX and endpoints are transmitted over a secure TLS connection instead of plain UDP or TCP.
| Setting | Description |
|---|---|
| Enable TLS | Toggle to enable or disable SIP-TLS transport. |
| SIP TLS Port | The port used for SIP signaling over TLS. Default: 5061. This field is read-only when TLS is disabled. |
| TLS Connection Method | The TLS protocol version to use. Default: TLS v1.2. |
Tip: Enabling TLS encrypts only the SIP signaling (call setup). To encrypt the audio stream as well, configure SRTP on the individual extension or trunk settings.
Session
The Session tab configures SIP session timers, which ensure that active calls are periodically refreshed to detect and release stale sessions (e.g., when a network failure prevents a normal call hangup).
Session Timer
| Setting | Description |
|---|---|
| Session Timer | Defines how the PBX handles session timer negotiation. Options: Supported (accept if requested by remote), Required (always require), Disabled (never use). |
| Session-Expires (s) | Maximum duration (in seconds) before a session must be refreshed. Default: 1800 (30 minutes). If no refresh is received, the call is terminated. |
| Min-SE (s) | Minimum session expiration time. Default: 90 seconds. The remote party cannot request a shorter interval. |
Session Parameter Configuration
| Setting | Description |
|---|---|
| Session Name | A custom name included in the SIP session description. Typically left empty. |
| Session Owner | A custom owner string included in the SIP session description. Typically left empty. |
QoS
The QoS (Quality of Service) tab defines the DSCP (Differentiated Services Code Point) and CoS (Class of Service) values for SIP, audio, and video traffic. These values tell network equipment how to prioritize different types of PBX traffic.
| Setting | Default | Description |
|---|---|---|
| ToS SIP | CS3 | DSCP value for SIP signaling packets. |
| ToS Audio | EF | DSCP value for audio (RTP) packets. EF (Expedited Forwarding) provides the highest priority. |
| ToS Video | AF41 | DSCP value for video (RTP) packets. |
| CoS SIP | 3 | 802.1p Class of Service for SIP signaling (Layer 2). |
| CoS Audio | 5 | 802.1p Class of Service for audio traffic. |
| CoS Video | 6 | 802.1p Class of Service for video traffic. |
Tip: QoS values are only effective if your network switches and routers are configured to recognize and act on DSCP/CoS markings. Contact your network administrator to ensure proper QoS policies are in place.
T.38
The T.38 tab configures the T.38 fax-over-IP protocol. T.38 provides more reliable fax transmission over IP networks compared to sending fax tones through a standard audio codec.
| Setting | Description |
|---|---|
| T.38 Max Bitrate | Maximum bitrate for T.38 fax transmission in bits per second. Default: 14400. |
| No T.38 Attributes in re-INVITE SDP | When enabled, T.38 attributes are not included in re-INVITE SDP messages. Enable this if you experience compatibility issues with certain fax gateways. |
| Error Correction Mode | When enabled, uses redundancy to correct transmission errors during fax. Recommended for networks with packet loss. |
Advanced
The Advanced tab provides additional SIP behavior settings for caller ID handling, SIP headers, and protocol options.
Incoming Caller ID / Number Retrieval
| Setting | Description |
|---|---|
| Get Caller ID From | Specifies which SIP header field to read the caller ID (name) from. Options: From (default), P-Asserted-Identity, Remote-Party-ID. |
| Get DID Number From | Specifies which SIP header field to read the called number from. Options: Invite (default), To. |
SIP Request Header
| Setting | Description |
|---|---|
| Application Agent | The User-Agent string sent in SIP requests. Default: Virtual-Call. |
| Internal Alert Info | Custom Alert-Info header value for internal calls. It can be used to trigger a distinctive ringtone on compatible phones. |
Other Options
| Option | Description |
|---|---|
| Support Request Message | Enable support for SIP MESSAGE requests (used for text messaging between extensions). |
| Inband Progress | When enabled, sends early media (ringback tone) in-band instead of using the 180 Ringing response. Useful for certain trunk providers. |
| Enable uaCSTA Connection | Enables uaCSTA (Computer Supported Telephony Application) for third-party call control integration. |
| Extension Forward with Diversion SIP Header | Includes a diversion header when forwarding calls, allowing the receiving party to see the original called number. |
| P Asserted Identity | Includes the P-Asserted-Identity header in outbound SIP requests to transmit the verified caller identity. |
| Call Connection Safeguard | Adds additional safeguards to prevent calls from being connected when one side has already hung up. |
Important: The advanced settings affect SIP interoperability with trunks and endpoints. Only change these if instructed by Virtual-Call support or if you have specific requirements from your SIP provider.
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