Configure Peer-to-Peer SIP Programs
Contents
SIP programs are like a normal IP program to configure, with two small differences; entering a SIP address and selecting SIP as the Session Protocol.
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Important Notes: Before you start program configuration please note: •You cannot edit a program when it is currently loaded in the codec. •Some drop-down menus and settings may be greyed out intentionally depending on features available. •Failover and SmartStream PLUS redundant streaming are not available with SIP connections. •To learn more about programs see the section titled About Program Dialing. |
1.Open the Java Toolbox Web-GUI and click the Programs 
symbol at the top of the screen to display the Programs panel.
2.Click the New Program button to open the wizard and:
•Click in the Program Name text box to name the new program.
•Select Mono/Stereo Peer-to-Peer, or if you want to use an existing program as a template, select this option. Then click Next.
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Important Notes: When you choose to use an existing program as a template, the new program inherits all the settings of the template program and you can adjust these settings as required by continuing through the program wizard. |
3.Enter a name for the Audio Stream and configure the codec to dial, answer or dial and answer. Then click Next.
Note: The following example will display how to configure a dial and answer program. If you want the codec to either dial or answer only, select the option and the wizard will automatically display screens to allow you to configure the codec correctly.
4.This audio stream connection in the wizard will allow the codec to dial. Enter the name of the connection in the text box, then click Next.
5.Follow the instructions on the right-hand side of the panel to configure the transport settings for the connection: Ensure that you select:
•IP as the Transport.
•SIP from the Session Protocol menu option.
Then click Next.
6.Configure the destination codec Address, then specify the network interface used to dial the connection, e.g. Primary (Ethernet port 1).
At this point you can click Save Program and save the program with default algorithm and jitter settings. Alternatively, click Next to confirm and specify algorithm and jitter settings for this connection and configure backup audio settings (recommended).
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Important Notes: •If your codec is registered with same SIP registrar as the destination codec then you only need to enter the SIP user name to dial successfully. •The default UDP audio port when using SIP for a peer-to-peer connection is 5004 in Tieline codecs. To contact a codec that is behind a firewall or NAT-enabled router, it is essential that this and all other relevant ports are open and forwarded to the other device. |
7.Click the drop-down arrows on the right-hand side of each active drop-down menu to adjust the Encoding, Sample rate or Bit rate parameters. Click Next to continue.
Click to configure:
•Auto Jitter Adapt and the preferred auto jitter setting using the drop-down arrow for Buffer Priority, or
•Fixed Buffer Level and enter the Jitter Depth, which must be between 12ms and 5000 ms depending on the algorithm you select.
Click Save Program to save all settings, or click Next to configure Auto Reconnect. If you click Save Program, select the check-box if you want to connect the program immediately, then click Finish.
9.The newly created program will be displayed in the Programs panel and in the Master panel. Dial the program by loading and connecting using the Master panel, or dial the program manually using the codec front panel.