SipWarp
From VoIP.ms Wiki
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Once launched, SipWarp displays its main telephone interface, which includes a dial pad and call controls. | Once launched, SipWarp displays its main telephone interface, which includes a dial pad and call controls. | ||
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If the application remains on '''Registering''', verify the SIP server address, username, password, and internet connectivity. | If the application remains on '''Registering''', verify the SIP server address, username, password, and internet connectivity. | ||
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'''Recommendation:''' Using a USB headset can help reduce echo and background noise. | '''Recommendation:''' Using a USB headset can help reduce echo and background noise. | ||
| - | + | : SipWarp-Audio-Settings.png|none|450px|SipWarp audio device configuration.]] | |
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== Step 6 - Configure Network Settings == | == Step 6 - Configure Network Settings == | ||
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Suggested filename: SipWarp-Network-Settings.png | Suggested filename: SipWarp-Network-Settings.png | ||
| - | [[File:SipWarp-Network-Settings.png| | + | [[File:SipWarp-Network-Settings.png|450px|SipWarp RTP and STUN configuration.]] |
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For additional details, refer to the [https://sipwarp.com/help/codecs.htm SipWarp Codec Documentation]. | For additional details, refer to the [https://sipwarp.com/help/codecs.htm SipWarp Codec Documentation]. | ||
| - | + | : [[File:SipWarp-Codecs.png|none|450px|Recommended audio codec configuration for SipWarp.]] | |
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== Step 8 - Configure Incoming Calls == | == Step 8 - Configure Incoming Calls == | ||
Revision as of 18:10, 23 October 2026
SipWarp
SipWarp is a free, lightweight SIP softphone for Windows that allows users to make and receive telephone calls directly from their computers using a compatible Voice over IP (VoIP) provider such as VoIP.ms.
Developed using Rust, C, and Slint, SipWarp offers a modern and resource-efficient desktop interface designed for everyday VoIP communications.
The application can operate in the Windows system tray, allowing users to remain available for incoming calls while working with other applications.
SipWarp supports several features, including:
- SIP-based incoming and outgoing calls.
- Support for multiple audio codecs, including G.711, G.722, and Opus.
- CardDAV contact synchronization.
- CSV and VCF contact importing and exporting.
- Manual and automatic call recording.
- Custom WAV ringtones.
- Configurable microphones, speakers, and headsets.
- Call history and contact management.
- Windows system tray operation.
Official Website: SipWarp
Supported Operating System: Windows
Application Type: SIP Softphone
Cost: Free
Prerequisites
Before configuring SipWarp with VoIP.ms, ensure that you have the following:
- An active VoIP.ms account.
- A SIP subaccount configured in the VoIP.ms Customer Portal.
- SipWarp installed on a Windows computer.
- A stable internet connection.
- A microphone and speakers, or a compatible headset.
- An active DID (telephone number) if you intend to receive incoming calls from the public telephone network.
Note: A DID is not required for outgoing calls. However, you must configure a valid and authorized outbound Caller ID.
Step 1 - Download and Install SipWarp
- Visit the official SipWarp website at sipwarp.com.
- Click Download for Windows.
- Download the latest available version.
- Install or launch SipWarp according to the instructions provided with the download.
- Open the application.
Once launched, SipWarp displays its main telephone interface, which includes a dial pad and call controls.
Step 2 - Create a VoIP.ms Subaccount
It is recommended to use a dedicated VoIP.ms SIP subaccount for SipWarp instead of configuring the primary account.
This allows you to manage the application's credentials, Caller ID, and calling permissions independently from other devices.
- Log in to the VoIP.ms Customer Portal.
- Navigate to Sub Accounts → Create Sub Account.
- Enter a descriptive username, such as
sipwarp. - Select SIP as the protocol.
- Select User/Password Authentication.
- Enter a strong and unique SIP password.
- Configure the appropriate Device Type and Caller ID Number.
- Review any required calling permissions.
- Save the subaccount.
Your complete SIP username follows this format:
123456_sipwarp
Where:
-
123456represents your VoIP.ms main account username. -
sipwarprepresents the name of your subaccount.
Important: The full SIP username, including the underscore and subaccount name, must be entered in SipWarp.
The SIP password is the password assigned to your subaccount, not the password used to access the VoIP.ms Customer Portal.
For more information, refer to VoIP.ms Subaccounts.
Step 3 - Select a VoIP.ms Server
VoIP.ms operates multiple Points of Presence (POPs) across North America and internationally.
For optimal performance, select a server geographically close to your location or the server already assigned to your DID.
For example:
montreal.voip.ms
Other examples include:
-
toronto.voip.ms -
newyork.voip.ms -
chicago.voip.ms
For the complete list of available servers, refer to VoIP.ms Servers.
Important: The VoIP.ms server used for SIP registration must match the POP assigned to your DID for incoming calls to reach your device correctly.
SIP Server Information
| Parameter | Value |
|---|---|
| SIP Server / Registrar | Your selected VoIP.ms POP |
| SIP Port | 5060 |
| Transport Protocol | UDP |
| SIP Username | 123456_sipwarp |
| Authentication Username | 123456_sipwarp |
| SIP Password | Your subaccount SIP password |
| SIP Domain | Same as SIP Server |
Note: VoIP.ms also supports alternative SIP ports for compatible configurations. The default UDP port 5060 is sufficient for this guide.
Step 4 - Configure the SIP Account in SipWarp
After creating your VoIP.ms subaccount, open SipWarp and configure the SIP registration settings.
- Open SipWarp.
- Click Account at the bottom of the application window.
- Enter your VoIP.ms SIP credentials.
- Configure the SIP server and connection settings.
- Click Save account.
- Wait for the registration status to change to Online.
Account Configuration
Use the following configuration as an example:
| SipWarp Field | Configuration | Description |
|---|---|---|
| Server | montreal.voip.ms | The VoIP.ms POP selected for your account. |
| Domain | montreal.voip.ms | Use the same hostname as the SIP server. |
| Username | 123456_sipwarp | Your complete VoIP.ms SIP subaccount username. |
| Password | Your SIP subaccount password | Password configured in the VoIP.ms Customer Portal. |
| Display Name | Your name or business name | Name used to identify the SIP account. |
| Port | 5060 or 0 | Port 0 allows SipWarp to use its default SIP port. |
| Transport | udp | Standard SIP transport for this configuration. |
Important: Replace the example values with your own VoIP.ms account information.
When registration succeeds, SipWarp displays:
Online
This indicates that the application has successfully registered with the VoIP.ms SIP server.
If the application remains on Registering, verify the SIP server address, username, password, and internet connectivity.
Step 5 - Configure Audio Devices
SipWarp allows you to select different audio devices for conversations and incoming call notifications.
To configure your audio devices:
- Open Settings.
- Select Devices.
- Choose the microphone you want to use.
- Select your preferred speaker or headset.
- Configure the device used for incoming call notifications.
- Save your configuration.
Audio Device Settings
| Setting | Description |
|---|---|
| Microphone | Selects the microphone used during calls. |
| Speaker / Headphones | Selects the device used to hear the other party. |
| Ring Device | Selects the device that plays incoming call notifications. |
| Ringtone File | Allows you to select a custom WAV ringtone. |
| Local DTMF | Plays a local keypad tone when pressing a digit. |
SipWarp allows you to use separate devices for the ringtone and call audio.
For example, incoming calls can ring through the computer speakers while conversations are handled through a USB headset.
Recommendation: Using a USB headset can help reduce echo and background noise.
- SipWarp-Audio-Settings.png|none|450px|SipWarp audio device configuration.]]
Step 6 - Configure Network Settings
SipWarp provides configurable RTP and STUN options to assist with SIP communications across networks using Network Address Translation (NAT).
Navigate to:
Settings → Network
Recommended Network Configuration
| Parameter | Recommended Value | Description |
|---|---|---|
| RTP Minimum Port | 10001 | Starting UDP port for RTP audio. |
| RTP Maximum Port | 20000 | Ending UDP port for RTP audio. |
| STUN | Enabled | Assists with identifying the public network address. |
| STUN Server | stun.voip.ms | STUN server recommended for VoIP.ms. |
| rport | Enabled | Helps SIP responses reach the correct network address and port. |
| DNS SRV | Disabled | Not normally required when using a specific VoIP.ms POP hostname. |
Understanding RTP
SIP signaling is used to establish and manage telephone calls. The audio itself is carried separately using RTP (Real-time Transport Protocol).
If an outgoing call connects but one or both parties cannot hear audio, the issue may be related to RTP connectivity.
Understanding STUN
STUN (Session Traversal Utilities for NAT) allows SipWarp to discover the public-facing address used by the network.
This can help resolve audio issues when the computer is located behind a router.
The recommended STUN server is:
stun.voip.ms
Note: STUN does not resolve every type of NAT or firewall issue.
Firewall and Port Forwarding
Port forwarding is generally unnecessary when calls are working correctly.
If calls connect but audio is missing, review the local firewall, router configuration, and RTP traffic.
Avoid exposing unnecessary ports to the public internet. Any firewall or port-forwarding changes should be limited to what is required for the installation.
For additional details, refer to the SipWarp Network Configuration Guide.
Step 7 - Configure Audio Codecs
SipWarp supports several audio codecs that determine how voice is encoded and transmitted during calls.
The application negotiates a compatible codec with the remote SIP endpoint based on the enabled codecs and their configured priority.
For VoIP.ms, G.711u (PCMU) is recommended as the first codec.
Recommended Codec Priority
| Priority | Codec | Description |
|---|---|---|
| 1 | G.711u (PCMU) | Recommended for regular telephone calls in North America. |
| 2 | G.722 | HD Call - Wideband audio for compatible SIP-to-SIP connections. |
Important: Both endpoints must support the negotiated codec.
G.722 may provide improved audio quality on compatible SIP calls. However, it is not generally available for ordinary calls to the public telephone network.
Opus support in SipWarp does not guarantee Opus interoperability with VoIP.ms.
Additional Codec Options
| Setting | Recommended Value | Description |
|---|---|---|
| VAD | Disabled | Prevents speech clipping caused by voice activity detection. |
| Echo Cancellation | Enabled | Helps reduce echo during conversations. |
| Opus Stereo | Disabled | Stereo audio is generally unnecessary for telephone calls. |
Other available codecs may be left disabled unless specifically required.
For additional details, refer to the SipWarp Codec Documentation.
Step 8 - Configure Incoming Calls
To receive incoming calls through SipWarp, a VoIP.ms DID must be routed to the SIP subaccount configured in the application.
- Log in to the VoIP.ms Customer Portal.
- Navigate to DID Numbers → Manage DID(s).
- Locate the DID you want to configure.
- Click the edit icon.
- Under Routing, select SIP/IAX.
- Choose the SIP subaccount associated with SipWarp.
- Verify that the DID's POP matches the SIP server configured in SipWarp.
- Save your changes.
Once the DID is properly routed and SipWarp is registered, incoming calls should be delivered to the application.
Important: If you intend to receive calls on multiple devices simultaneously, use separate SIP subaccounts and configure a VoIP.ms Ring Group.
Registering multiple devices with the same SIP credentials is not recommended because it can cause registration conflicts.
Step 9 - Configure Outgoing Calls
To place an outgoing call:
- Open SipWarp.
- Enter the destination telephone number using the dial pad.
- Click Call.
- Wait for the call to connect.
North American Dialing Format
For calls to telephone numbers within the North American Numbering Plan, the following dialing format is recommended:
1 + Area Code + Telephone Number
For example:
13125551212
For international calls, consult the VoIP.ms Dialing Codes documentation.
Outbound Caller ID
The outbound Caller ID is generally configured through the VoIP.ms Customer Portal.
When using a dedicated SIP subaccount:
- Navigate to Sub Accounts → Manage Sub Accounts.
- Edit the SipWarp subaccount.
- Locate the CallerID Number setting.
- Select or configure an authorized outbound Caller ID.
- Save your changes.
SipWarp may also send Caller ID information from the application, but this requires the appropriate configuration on the VoIP.ms subaccount.
Important: Only telephone numbers you own or are authorized to use should be configured as outbound Caller ID.
Step 10 - Test Your Configuration
After completing the configuration, perform the following tests to confirm proper operation.
| Test | Expected Result |
|---|---|
| SIP Registration | SipWarp displays Online. |
| Outgoing Call | Calls connect successfully. |
| Incoming Call | The assigned DID rings SipWarp. |
| Two-Way Audio | Both parties can hear each other. |
| DTMF | Keypad tones are properly recognized. |
| Caller ID | The authorized outbound number is displayed. |
| Audio Devices | The microphone and headset operate correctly. |
VoIP.ms Test Numbers
VoIP.ms provides test numbers that can help verify your configuration.
| Number | Description |
|---|---|
| 4443 | Echo test. Allows you to verify audio transmission and reception. |
| 822 | Caller ID test. |
The echo test is useful for identifying missing audio, one-way audio, or microphone-related issues.
Additional Features
Contact Management
SipWarp includes contact management functionality.
Contacts can be stored locally and imported or exported using common file formats.
Supported formats include:
- CSV (Comma-Separated Values)
- VCF (vCard)
These features allow users to migrate contacts between compatible applications.
CardDAV Synchronization
SipWarp supports CardDAV, a protocol used to synchronize contact information between devices and applications.
To configure CardDAV:
- Open Settings → CardDAV.
- Enter the CardDAV server URL.
- Enter the username and password associated with the CardDAV service.
- Click Test to verify connectivity.
- Save the configuration.
Example CardDAV server URL:
https://contacts.example.com
Note: CardDAV credentials are separate from your VoIP.ms SIP credentials.
For additional details, refer to the SipWarp CardDAV Documentation.
Call Recording
SipWarp supports manual and automatic call recording.
To record an individual call, use the REC button during an active conversation.
To enable automatic call recording:
- Open Settings → Devices.
- Locate Record all calls automatically.
- Enable the option.
- Save your changes.
Important: Users are responsible for complying with applicable call recording, privacy, and consent requirements in their jurisdiction.
Custom Ringtones
SipWarp allows users to configure a custom WAV ringtone.
- Open Settings → Devices.
- Locate Ringtone File.
- Select the desired WAV file.
- Save your configuration.
If no custom ringtone is configured, SipWarp uses its built-in ringtone.
Background Operation
SipWarp can continue running in the Windows system tray after its main window is closed.
This allows the application to remain available for incoming calls.
To reopen SipWarp, double-click its system tray icon.
To completely close the application, right-click the system tray icon and select Exit.
Launch at Windows Startup
To start SipWarp automatically when signing in to Windows:
- Open Settings → App.
- Enable Run at startup.
- Save the configuration.
Troubleshooting
The following table lists common issues and recommended troubleshooting steps.
| Issue | Possible Cause | Recommended Solution |
|---|---|---|
| Registration remains on Registering | Incorrect SIP credentials or server. | Verify the SIP username, password, server hostname, and port. |
| Cannot receive incoming calls | Incorrect DID routing or POP mismatch. | Verify that the DID routes to the correct subaccount and uses the same POP. |
| Outgoing calls fail | Incorrect dialing format or Caller ID configuration. | Verify the destination number, Caller ID, and account balance. |
| No audio | Incorrect audio device or blocked RTP traffic. | Verify the microphone, speaker, firewall, and RTP settings. |
| One-way audio | NAT or firewall issue. | Review STUN configuration and RTP connectivity. |
| Poor audio quality | Network instability or audio device issues. | Check latency, packet loss, and audio device settings. |
| DTMF not working | Incompatible DTMF configuration. | Verify the DTMF settings on both the subaccount and the application. |
| Registration disconnects | Network instability or SIP connectivity issue. | Check internet connectivity, firewall rules, and SIP registration settings. |
| No incoming ringtone | Incorrect ring device. | Verify the Ring Device under Settings → Devices. |
Verify Registration Status
You can verify whether your subaccount is registered through the VoIP.ms Customer Portal.
If SipWarp displays Online but incoming calls do not arrive, review the DID routing configuration and POP assignment.
Verify Firewall and NAT Settings
If SIP registration is successful but audio is missing, investigate possible firewall or NAT issues.
The problem may involve RTP traffic rather than SIP registration itself.
Review the network configuration before making changes to firewall rules.
Security Recommendations
For improved security, the following practices are recommended:
- Use a dedicated SIP subaccount for SipWarp.
- Configure a strong and unique SIP password.
- Avoid sharing SIP credentials between devices.
- Restrict international calling when it is not required.
- Enable appropriate security restrictions in the VoIP.ms Customer Portal.
- Keep SipWarp and Windows updated.
- Protect locally stored SIP account information.
- Use encrypted SIP signaling and media only when both the application and provider are configured to support the selected encryption methods.
Note: VoIP.ms supports SIP-TLS and SRTP. However, encrypted call functionality must be confirmed as compatible with the installed SipWarp version before enabling encryption on the VoIP.ms subaccount.
Enabling mandatory encrypted SIP traffic on the subaccount without a compatible client configuration may prevent successful registration or calling.
For more information, refer to VoIP.ms Call Encryption.
Local Application Data
SipWarp stores its application configuration and related data in the following Windows directory:
%APPDATA%\SipWarp
This directory may remain on the computer after the application is uninstalled.
If removing SipWarp permanently, review this directory for any configuration files, contacts, or other application data that should be backed up or deleted.
