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Tieline G6 Codec SIP Compatibility v1
© Tieline Pty. Ltd. 2021
Then click
Next
.
7. Configure the destination codec
Address
if you are dialing peer-to-peer, then specify the
network interface used to dial the connection, e.g.
Primary
(Ethernet port 1). Enter the
name of a registered SIP account if you are using a SIP server to establish a connection. If
you wish to dial from one of the codec's registered accounts, then enter the account name in
the
Account
field
using
the
format
accountname@sipserverdomain,
e.g.
. In this configuration the account interface will be used rather
than the specified
Via
, e.g. if the account is using
SIP2
and this is configured to use LAN2,
then the call will proceed using LAN2. If you do not wish to use an account for the dial then
leave the
Account
field blank and select the required interface. Note: the interface must be
associated with either
SIP1
or
SIP2
for the call to be able to proceed.
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).
Important Notes:
·
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.
·
Tieline codecs automatically add "
sip:
" to the address you enter in the
Address
field
when dialing, so it's not necessary to add this.
·
Enter the IP address or SIP URI, then a full colon and the session port number to change
the session port from the default setting 5060.
8. 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.
