From SIPfoundry sipx, The Open Source SIP PBX for Linux - Calivia

Contents 1 linphone 0.12 1.1 Network 1.2 Sound device 1.3 SIP 1.4 Codecs 2 linphone 1.0

[edit] linphone 0.12

[edit] Network

NAT traversal options

Version 0.12 does not support STUN.

• Enable and add the external IP address of your firewall.

RTP properties

RTP traffic will have to be NATted statically to a configurable port.

• Choose a port and make sure this port gets NATted to your machine.

[edit] Sound device

Sound properties

Do not use OSS sound.

[edit] SIP

SIP Port

Only change if this port is already used by another application.

Identity

Enter your SIP identity

Remote services

• Choose Register to register with sipX
• Server address
• Address of record:
• Use as outbound proxy

[edit] Codecs

Audio Codecs

Will be enabled or disabled based on selected bandwidth.

[edit] linphone 1.0