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.

Use the Softphones tab in the Administrator to configure SIP parameters. In order to set up the Softphone in the Administrator:

Add at least one Softphone component in to the design

Have the SIP or IP/PBX information ready to enter

Know which Audio codecs you are going to use

Know which LAN port on the Core you are going to use to connect to the SIP device

To Set up the Softphone

Open the Q-SYS Administrator. You may use the stand-alone application, or open it from inside Q-SYS Designer.

Click the Softphones tab.

Under Softphones in the Administrator workspace, double click one of the Softphone entries. (The number of entries and their names equals the number (maximum: Core 110f = 2, all other Cores = 64) of Softphone components you have in your design Inventory, and what you have named them.) The Edit Softphone dialog opens.

Enter the required information as listed in the table below.

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
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.	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)		User Input

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

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.

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)

User Input

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.	LAN A, LAN B, Aux A, or Aux B¹
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
DTMF Payload Type	The DTMF Payload Type 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. The DTMF Payload Type used by default 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: Currently 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
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
1. Not all Cores have an Aux B port.

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.

LAN A, LAN B, Aux A, or Aux B¹

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

DTMF Payload Type

The DTMF Payload Type 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.
The DTMF Payload Type used by default 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: Currently 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

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

1. Not all Cores have an Aux B port.

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