Softphones (Administrator)

The Q-SYS Softphone gives you the ability to connect to a Voice-over-IP telephone system (IP-PBX or SIP-based devices. The Softphones tab in the Administrator facilitates registration with IP-PBX systems (such as Cisco CUCM, FreeSwitch, etc.). In addition, you can use the Softphone component in unregistered mode, which allows making ad-hoc IP-to-IP calls and connecting to other non-registered SIP-compatible equipment.

NOTE: The maximum number of Softphones that can be placed in a design is: Core 110f = 4; all other Cores = 64.

Controls

Individual Softphone Parameters

Control	Function	Default / Range
Name	This is the name given to the Softphone Component in the Properties. This displays automatically in the Administrator's Softphone list.	Softphone-1 through n / User Input
User Name	This is the number or name you use to call the Softphone. Registered: The SIP or IP-PBX provider must know, and typically supplies this User Name. This along with the Password identifies your Softphone to the SIP or IP-PBX provider. Typically, in this case, it would be the extension number. Unregistered: This can be almost anything, for example, the local part (before the @ sign) of an email address.	User input
CID Name	Enter a name you wish to use for your caller ID.	User input
Proxy	The IP address or resolvable name of the SIP provider, IP-PBX, or SIP gateway.	User input
Backup Proxy (Optional)	The IP address or resolvable name of the SIP provider, IP-PBX, or SIP gateway to use if a connection to the primary proxy cannot be established.	User input
Transport	Select the protocol you want to use, TCP, UDP. or TLS ¹.	UDP / TCP / TLS
Register With Proxy	This pull-down list offers the choice to register the Softphone with the proxy or not.	No / Yes
Authentication ID	This field is available when Register With Proxy is set to Yes. The SIP or IP-PBX provider supplies this information. The Authentication ID is masked for security.	User input
Password	This field is available when Register With Proxy is set to Yes. The SIP or IP-PBX provider supplies this information.	User input
Domain (Optional)	Enter the Domain name of the PBX's network if required.	User Input
1. TLS (Transport Layer Security) is a protocol that runs over TCP and provides end-to-end security for SIP signaling by encrypting SIP messages that are exchanged between the Q-SYS SIP Softphones and far end SIP endpoints or PBXs.

Global Parameters

Control	Function	Default / Range
Core Interface	The network interface used for all SIP traffic. This port must have a route to the SIP device to which you want to connect. Note: The Softphone does not support usage within the 169.254.x.x IP address range.	LAN A, LAN B, Aux A, or Aux B²
SIP Port	This is the listening UDP port of Q-SYS Softphones. Allows adjustment of the SIP Listen port from the perspective of Q-SYS. Outbound calls will always be 5060.	Default = 5060
Enable DTMF INFO	Enables DTMF to be sent as SIP INFO Requests. When enabled, DTMF will not be sent via RFC 2833.	No / Yes
RFC2833 DTMF Type	This number is the ‘RTP Event’ Payload Type Number that indicates the transmitted packet contains DTMF digits. The entry must be in the dynamic payload number range of 96-127. Do not specify a number below 96. The default number on Q-SYS Softphones is 101. On the remote SIP device, RX and TX DTMF Payload Types must match this value. Some SIP devices can specify separate RX and TX DTMF Payload Types and some, as does Q-SYS, cannot. On the systems that cannot have separate values, RX and TX DTMF Payload Types must both match this value.	Default = 101 Range = 96 to 127
Enable Logging	Select Yes to enable SIP logging. This log can be used by QSC support to troubleshoot problems you may be experiencing.	No / Yes
Enable STUN	The STUN (Simple Traversal of UDP through NATs) feature allows you to communicate with a cloud-based PBX. STUN is a protocol that finds the public IP of a service (Q-SYS) that is behind a firewall or network address translator. A STUN packet is sent to a publicly accessible STUN server, which responds with the public-to-private IP address. Once Q-SYS receives the public address it uses that address to communicate with a cloud-based PBX. Note: If you enable STUN for one softphone, all softphones in your Q-SYS design will use STUN.	No / Yes
STUN Server (Optional)	The Q-SYS default for the STUN server is "stun.freeswitch.org". If you want, you can enter a different STUN server. Free STUN servers are listed at https://gist.github.com/zziuni/3741933	User Input
Enable SRTP	SRTP (Secure RTP) is a protocol for encrypting the RTP voice / media payload.	No / Yes
Audio Codecs	This is a list of the supported Audio Codecs ordered by preference of connection. You must check the Codecs you want to be available for use. To change the order, select a Codec you want to move, then use the up and down arrows to move the selected Codec,	User selection
2. Not all Cores have an Aux B port.

Softphones (Administrator)

1. TLS (Transport Layer Security) is a protocol that runs over TCP and provides end-to-end security for SIP signaling by encrypting SIP messages that are exchanged between the Q-SYS SIP Softphones and far end SIP endpoints or PBXs.

2. Not all Cores have an Aux B port.