VoIP CONFIGURATION FOR GPON
VoIP (voice over IP) is the transmission of voice and multimedia content over Internet Protocol (IP)
networks. VoIP is enabled by a group of technologies and methodologies used to deliver voice
communications over the internet, enterprise local area networks or wide area networks.
How does VoIP work?
VoIP encapsulates audio via a codec into data packets, transmits them across an IP network and de-
encapsulates them back into audio at the other end of the connection. VoIP endpoints include dedicated
desktop VoIP phones, softphone applications running on PCs and mobile devices, and WebRTC-enabled
browsers.
By eliminating the use of circuit-switched networks for voice, VoIP reduces network infrastructure
costs, enables providers to deliver voice services over their broadband and private networks, and allows
enterprises to operate a single voice and data network.
Step 1: Creating Service Profile.
1. Go to “Service Profile” option under “GPON” section.
2. Click on “Add Service Profile”.
Once the service profile is created, it is displayed on service profile page as shown below:
Step 2: Creating Connections for the ONTs.
1. Go to “Connections” option under “GPON” section.
2. Click on “Add New Connection”.
Note: The above figure is for the first ONT. The “Connections” should be created for each ONT
connected to the OLT with same “Service Profile”.
[ Here, Two ONTs have been used. Each Phone connected to “Phone 1 Port” of both the ONTs.]
Once the “Connections” are created, it is displayed on “Connections” page as shown below:
Step 3: Configuring “VOIP & IP Details” for ONTs.
1. Go to “GPON Ports”.
2. Click on “Port _GPON-1-x-y”. [ Here, x is slot number and y is port number where ONTs are
connected.]
3. Click on “View ONTs On: Port GPON-1-x-y.
4. Select the ONT to be configured.
5. Go to “VOIP & IP Details” as shown below.
6. Configure the parameters carefully for ONTs.
Phone 2 Port
Phone 1 Port
Here, “Proxy Address”, “Registrar IP Address” & “Outbound Proxy IP Address” refers to the IP
Address of the SIP Server.
ONT IP Details refers to the IP address of the ONT, Subnet mask & Gateway.
Note 1: Both SIP Server and ONT should have Internet Connectivity to make phone calls across ONTs.
Note 2: In Software build a42_42_1, the first Phone number refers to the “Phone 2 Port” on ONT and
second Phone number refers to the “Phone 1 Port” on ONT.
7. Configure the parameters for number of ONTs connected accordingly.
Phone 2 Port
Phone 1 Port
Step 4: Configure Ethernet Switching Parameters as shown below;
1. Go to “Port Configuration” option under “Service Switch” of “L2 Services”.
2. Click on “edit” of “Edit Switching Parameters” column.
Step 5: Creating Flow Point Template.
1. Go to “Flow Point Templates” option under “Services Provisioning” of “L2 Services”.
2. Click on “Provision a new FlowPointTemplate”.
3. Configure the parameters as shown below:
Once the “Flow Point Template” is created, it is displayed on “Flow Point Templates” page as shown
below:
Step 6: Creating Eline Service.
1. Go to “ELINE Services” option under “Services Provisioning” of “L2 Services”.
2. Click on “Add new ELINE Service”.
[ Here, Physical interface ETH-1-2-2 is mapped to logical interface BPETH-1-3-206. Since ONTs are
connected to 6th port of 3rd slot of PON card, BPETH-1-3-206 is to be selected in this scenario. It might
vary as per physical connection made.]
Once, ELINE Service is created, it is displayed on “ELINE Services” page as shown below:
Session Initiation Protocol (SIP):
SIP is the Session Initiation Protocol. Session Initiation Protocol (SIP) is a signaling protocol used for
initiating, maintaining, modifying and terminating real-time sessions that involve video, voice,
messaging and other communications applications and services between two or more endpoints
on IP networks. In IP and traditional telephony, network engineers have always made a clear distinction
between two different phases of a voice call. The first phase is "call setup," and includes all of the
details needed to get two telephones talking. Once the call has been setup, the phones enter a "data
transfer" phase of the call using an entirely different family of protocols to actually move the voice
packets between the two phones. In the world of VoIP, SIP is a call setup protocol that operates at the
application layer.
SIP can run over IPv4 and IPv6 and it can use either TCP or UDP. The most common implementations,
though, use IPv4 and UDP. This minimizes overhead, thereby speeding performance.
SIP features:
The SIP communications protocol determines five attributes when establishing and terminating
multimedia sessions:
• User location
• User availability
• User capabilities
• Session setup
• Session management
Step 7: Installing “Brekeke SIP Server” in the laptop & modifying some settings for the installed
SIP Server Admintool as shown below:
NETWORK DIAGRAM:
Here, ETH-1-2-2 of CEF7G card which is inserted in slot 2, is connected to Switch Port. The Gateway
in this scenario is 192.168.222.1 which is assigned to the interface f0/0 of the router (for example).
To enable VoIP service, both ONT and SIP server should communicate to each other, which is possible
when there is an Internet Connectivity.
Once, all the SIP Settings are modified correctly and internet connectivity is available, we will be able to
view the Registered clients under the “Registered Clients” tab as shown below:
VoIP Service Status and Activation Status of ONTs can be verified by logging into ONT and using the
command “voice show”. As shown in above figure, “VoipServiceStatus” is “Up” when there is an
internet connectivity between ONTs and SIP Server. Otherwise, “VoipServiceStatus” shows “Disabled”.
VERIFICATION:
As shown in above figure,
When the phone is ringing, it is shown under “Status” tab of “Active Sessions” as “Ringing”.
If the conversation is going on, it is shown under “Status” tab of “Active Sessions” as “Talking”.
