Started by BCITMike, March 08, 2013, 12:20:34 AM
VoIP Unlock Key CredentialsSupportposted this on Feb 07 12:53Version 1.01The Freephoneline VoIP Unlock Key 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 SettingsSIP Server: voip.freephoneline.caAlternative SIP Server:voip2.freephoneline.caTransport: UDPPort: 5060Notes: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 secondsRecommended SettingsSTUN/ICE:DisableNAT Mapping Enabled:YesNAT 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 PacketFor Linksys/Cisco devices, use 'Nat Keep Alive Msg' = $NOTIFY or $PINGNever use REGISTER as your Keep Alive messageKeep Alive Interval: 20 seconds* *Audio may be affected if this value is adjustedNotes: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.Freephoneline does not offer STUN server. However, you may use a public one if your wish.RTP SettingsSupported Codecs: g711-u (uLAW) and g729Suggested 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 networkIf 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.