SIP Phone System by Tragofone
Why SIP has sparked interest in the telephony market:
It is an open standard.
Since it is text-based, anyone can read a SIP message that is passed between two endpoints in a SIP session. The protocol is flexible.
It allows users to manage and modulate every call that enters their system.
SIP is content-agnostic. It is used for establishing sessions for voice, video, messaging applications, and file transfers.
Tells the protocol whether the party that is being called is available to take it.
Connects a call from one user to the other.
Tragofone uses APNS to send push notifications to iOS devices and Google FCM for Android phones.
Determines the media that will be used for the call.
Running of the call session. For example: transferring and ending a call.
How SIP powers a VoIP session call flow
When a user agent launches, it has to register with a SIP server so it can be found by other users. This happens by using the SIP Register request message.
Establishing a call:
SIP invite request: 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.
SIP response 100: A message is sent from the SIP server to the end user’s number in order to confirm the invite request.
SIP Response 180: This message is a confirmation that the invite has been received and the user agent is alerting the user.
SIP Response 200: 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.
Terminating a call:
When a user ends a call, a SIP 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.
How SIP adds value to VoIP
Provides crystal clear images in long-distance video conference calls.
Easy way to send files and documents.
Builds on VoIP’s capabilities by adding video and data transfer potential to it.
Not only does SIP set up the initiation of contact between devices, but it also allows
users to add more parties to a call, or switch between communication methods.
How to get your Business ready for SIP Calling
Robust network infrastructure:
Verify that your existing hardware, routers, and wireless network is capable to handle Quality of Service (QoS) settings and supports a SIP-based environment.
Bandwidth & VoIP-compatible:
Add higher bandwidth than your current data connection for a smooth transition. Upgrade to VoIP-compatible telephone devices. Some traditional phones can be upgraded with VoIP adapters
You may need to upgrade your PBX system to make it compatible with SIP calling. It is also advisable to switch to a full cloud-based communications solution to keep the transition smooth.
SIP Benefits for Businesses
Make long-distance calls at the price of local calls.
SIP calling makes it easy to route to mobile phones or other workstations.
Your entire suite of communications—internal and external—can be combined into one easy-to-use, computer-based system.
With SIP calling, your business can hire a remote workforce that is not tied to a desk.
Add and manage SIP lines seamlessly through a dedicated web-based portal.
There are several tools available to monitor and resolve issues in quick time.
SIP can give you significant returns on a relatively small investment.