Mayah's range of IP codecs is fully compatible with sip.audio. It includes OPUS, our preferred codec for interoperability.

1. Log in to the web interface for your Mayah unit using its IP address.
2. Select 'Setup' and click on 'Profiles', then 'New'.
   Name: sip.audio
   Algorithm: Opus
   S. Rate: 48 KHz
   Bitrate: 128 KBit/s
   Mode: Mono
   Resolution: 16 Bit
   Inband FEC: 5
   Follows: Remote
   IMUX: No
   Save
3. Click the sip.audio profile and then 'Apply'.
4. From the menu, select SIP:
   Keep alive: 600
   Use STUN: on
   Account Registrar: sip.audio
   Account Username: SIP Account Username (For example, enter fmstation.tieline for fmstation.tieline@sip.audio)
   Account Password: SIP Account Password
   STUN server: stun.sip.audio
   Proxy: proxy.sip.audio
   Account active: on
   OK

From the menu, select 'Direct':
Mode: normal
Protocol: SIP
Direction: sendrcv
Address: The SIP address you wish to call.
Click 'Dial' button.

With more than one Mayah codec on the same network, it's necessary to set a different SIP port on each unit. e.g. 5060, 5061, 5062...

If the unit has a second Ethernet interface that is not in use or does not have Internet access, disable it.

If there is no or one-way audio, make sure the correct channels are faded up on the internal mixer.

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