The LifeTimeLine VoIP provides sip credentials that can be used to configure any SIP client to work with the Freephoneline service. Please review the following Freephoneline guidelines to set up your SIP client. Failure to follow the required guidelines will result in account suspension followed by a notification email. Once your configuration adheres to the guidelines, service will be restored.

These guidelines may change over time - if and when they are changed freephoneline users will be notified at the email address used for account login a minimum of 7 days prior to the changes being required. Please ensure your contact information is accurate by visiting your account profile on freephoneline.ca.

Required Settings

SIP USERNAME & PASSWORD PROVIDED IN CLIENTAREA

SIP Server:	voip.freephoneline.ca
Alternative SIP Server:	voip2.freephoneline.ca
Transport:	UDP
Port:	5060

Rogers Customers with Hitron CGN2/CGN3 Modem,
or other users having issues connecting on port 5060:

SIP Server:	voip4.freephoneline.ca:6060
Transport	UDP

Notes:

• It is always best to use the DNS name for your SIP server as our infrastructure is always expanding/changing/being maintained. The IP addresses which you register to will change over time.
• Use of Fongo SIP Servers that are not listed in this document will result in your account being suspended.

Registration Interval:	3600 seconds (1 hour)
Registration Expiry:	3600 seconds (1 hour)
Failed Registration Re-Try Interval:	120 seconds

Recommended Settings

STUN/ICE:	Enable : STUN stun.unmetered.io
NAT Mapping Enabled:	Yes
NAT Traversal:	Enable sending Keep-Alives only: on Grandstream HT-701 ATAs this setting is “no, but send keep-alive”
Keep Alive Message:	NOTIFY or a UDP PING Packet For Linksys/Cisco devices, use ‘Nat Keep Alive Msg’ = $NOTIFY or $PING Never use REGISTER as your Keep Alive message
Keep Alive Interval:	20 seconds (Audio may be affected if this value is adjusted)

Notes:

• The above settings can be used to configure your SIP client to function in common home network configurations. Since there a thousands of home network configurations, it is impossible for us to provide a single set of parameters that will always work. As a VoIP Key purchaser, it’s expected that you have knowledge of your network and how to configure your SIP client properly.

RTP Settings

Supported Codecs:	g711-u (uLAW)  and g729
Suggested RTP Packet size (psize):	0.020  - This ensures audio packets every 20 milliseconds, achieving better quality (trade-off: bandwidth)

Notes:

• The above settings are used by your ATA to determine how the audio will be encoded/decoded across the Fongo network.

Additional Information for users with multiple SIP clients on their network

If you use multiple VoIP providers or SIP clients, including Dell Voice or Fongo Mobile on the same network you may encounter issues if your router does not support UPNP.