The 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 Settings
| SIP Server: |
voip.freephoneline.ca |
| Alternative SIP Server: |
voip2.freephoneline.ca |
| Transport: |
UDP |
| Port: |
5060 |
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: |
Disable |
| 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.
-
Freephoneline does not offer STUN server. However, you may use a public one if your wish.
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.
great post
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