Network Related

These options set important network related values regarding NAT, monitoring and security.

Transport: Type of transfer protocol that will be used on CloudPBX.

UDP (User Datagram Protocol) - With UDP, computer applications can send messages, in this case referred to as datagrams, to other hosts on an Internet Protocol (IP) network without prior communications to set up special transmission channels or data paths.

TCP (Transmission Control Protocol) - provides reliable, ordered, error-checked delivery of a stream of octets between programs running on computers connected to an intranet or the public Internet.

TLS (Transport Layer Security) - cryptographic protocol that provide communication security over the Internet.[1] They use asymmetric cryptography for authentication of key exchange, symmetric encryption for confidentiality, and message authentication codes for message integrity.

WebRTC Enabled: Navigate to Settings > Protocols > SIP and make sure TLS is set to “Yes”. Under Extensions > Extension > Edit screen, make sure option WebRTC Enabled is set to Yes. For Allowed Codecs make sure “Opus” is enabled and set.

NOTE: In order for WebRTC to function properly, first thing is to make sure SSL certificates are set properly. Go to Setup Wizard > SSL Certification page and make sure to upload a valid certificate or to use automatic Let’s Encrypt service.

Encryption: This option enables or disables encryption in CloudPBX transport.

NAT (Network Address Translation) Set the appropriate Extension - CloudPBX NAT relation.

If extension 1000 is trying to register with the CloudPBX from a remote location/network and that network is behind NAT, select the appropriate NAT settings here:

• • • yes - Always ignore info and assume NAT

    • no - Use NAT mode only according to RFC3581

    • Default (rport) - this setting forces RFC3581 behavior and disables symmetric RTP support.

    • Comedia RTP - enables RFC3581 behavior if the remote side requests it and enables symmetric RTP support.

Direct Media:

• • • No - this option tells the Asterisk to never issue a reinvite to the client

    • Yes - send reinvite to the client

    • No NAT only - allow reinvite when local, deny reinvite when NAT

    • Use UPDATE - use UPDATE instead of INVITE

    • No NAT, Update - use UPDATE when local, deny when NAT

Direct RTP setup: Here you can enable or disable the new experimental direct RTP setup. Setting this value to yes sets up the call directly with media peer-2-peer without re-invites. Will not work for video and cases where the callee sends RTP payloads and fmtp headers in the 200 OK that does not match the callers INVITE. This will also fail if directmedia is enabled when the device is actually behind NAT.

Qualify: Timing interval in milliseconds at which a 'ping' is sent to the UAD/Phone or trunk, in order to find out its status(online/offline). Set this option to '2500' to send a ping signal every 2.5 seconds to the UAD/Phone or trunk. Navigate to 'Monitor: Extensions' or 'Monitor: Trunks' and check the 'Status' field. In CloudPBX 5.0 'Qualify' is set to 8000 by default.

Host: Set the way the UAD/Phone registers to CloudPBX. Set this field to 'dynamic' to register the UAD/Phone from any IP address. Alternately, the IP address or hostname can be provided as well.

Default IP: Default UAD/Phone IP address. Even when the 'Host' is set to 'dynamic', this field may be set. This IP address will be used when dynamic registration could not be performed or when it times out.

NOTE: UAD/Phone must be on static IP address.

Use RTP source address for T.38 packets (1.2) Use the source IP address of RTP as the destination IP address for UDPTL packets if the nat option is enabled. If a single RTP packet is received Asterisk will know the external IP address of the remote device. If port forwarding is done at the client side then UDPTL will flow to the remote device.