Its open standard allows clients to achieve vendor-neutrality goals for communication needs.
Unlike traditional telephony protocols, SIP powers unified
communications by integrating voice, video, messaging, and file transfers into a single,
scalable platform.
SIP’s text-based protocol makes it highly transparent and easy to troubleshoot.
Its content-agnostic design ensures it can handle diverse media types.
Tragofone’s push notifications feature enables you to be active 24/7 on your mobile device using the app.
Tragofone supports HD voice and 4K video calls with features like call transfer, voicemail, call recording, and 3-way conferencing for smooth professional communication.
Tragofone integrates with tools like Zoho, HubSpot, PortaSwitch and Microsoft Teams. Whether you are managing leads or handling SIP infrastructure, integrations are ready to plug into your ecosystem.
You can fully rebrand Tragofone, change its name, logo, colors, and UI to match your business. From interface language to call settings, everything is customizable for a branded experience.
Stay connected through one-to-one or group chat, SMS, and MMS. You can sync contacts, share files, and track availability with real-time presence, all from one place.
Using SIP integrations with Push Call Server (PCS), Tragofone ensures desktop and mobile devices receive call alerts in real time, even when the app is closed.
Tragofone simplifies large-scale deployments with SIP-based auto-provisioning. Admins can configure settings remotely via a centralized dashboard.
Tracks SIP-level metrics like session quality, network type, call abandonment, and device-specific logs to help businesses monitor and optimize call performance.
SIP-enabled contact syncing allows Tragofone to integrate with enterprise directories and LDAP, so users have unified access to their business contacts.
Enables SIP-based call management capabilities such as transferring calls, putting them on hold, and VoIP conferencing calls—crucial for business-grade telephony.
Tragofone protects your conversations using TLS, SRTP, DTLS, and secure WebSocket protocols.
Tragofone supports simultaneous multi-device login, syncing SIP sessions and call logs in real time across desktop and mobile.
Tragofone allows SIP rule-based forwarding of calls and supports emergency number routing as defined in enterprise SIP policies.
Tragofone uses APNS to send push notifications to iOS devices and Google FCM for Android phones.
Tells the protocol whether the party that is being called is available to take it.
Determines the media that will be used for the call.
Connects a call from one user to the other.
Running of the call session. For example: transferring and ending a call.
When a user agent launches, it has to register with a VoIP SIP server so it can be found by other users. This happens by using the SIP Register request message.
The SIP invite initiates an attempt to establish a call. This is sent from the caller to
the SIP server where it finds the call receiver’s number. Once the number is located,
the invite is forwarded.
A message is sent from the SIP server to the end user’s number in order to confirm the
invite request.
This message is a confirmation that the invite has been received and the user agent is
alerting the user.
A 200 response is sent to confirm a call when it is answered by the end-user. This
message results in an exchange and negotiation of VoIP call parameters. SIP Ack Request:
The caller confirms the call by sending an Ack request to the number called.
The VoIP call is transmitted between the user agents using RTP (real-time transport protocol), which is also used for sending audio and video data over IP networks.
When a user ends a call, a SIP softphone client Bye request is sent. The call can be ended from either side of the line. The termination is confirmed by the other user agent with a SIP 200 status code response.
Reduces infrastructure costs by replacing traditional phone lines with SIP trunks
Supports voice, video, messaging, and presence information in a single platform
Enables call transfer, voicemails, auto-attendants, conferencing, and CRM integration
Supports encryption protocols like TLS, SRTP and WebSocket
Adapts to emerging business communication technologies like WebRTC and IoT
Verify that your existing hardware, routers, and wireless network are fully capable of efficiently handling Quality of Service (QoS) settings and support a SIP-based environment.
Ensure your network has sufficient bandwidth to handle VoIP traffic alongside existing data usage. Upgrade to a VoIP phone system, or some traditional phones can be upgraded with (ATAs).
To support SIP calling, you may need to upgrade your existing PBX with SIP trunking. It is also advisable to switch to a cloud-based communication solution to keep the transition smooth and future-proofing.
Get quick answers to common questions about our VoIP softphone solutions and features