SIP Config - Advanced View

Session Initiation Protocol (SIP) Settings

SIP is an application layer protocol used to establish, modify and terminate VoIP calls. The following image shows SIP functions available from the VoIP management website.

A description of each field/function follows. Note that different vendors use different nomenclature with regard to naming conventions: see Vendor Naming Conventions.

Proxy Vendor - allows the user to choose the entry that matches the phone system the VoIP Phone is integrating with.
Extension/SIP User Name/Line - the alphanumeric string that identifies the VoIP extension on the network. It is the number or string to dial to reach this extension.

Authentication User Name (Extension) - the credentials needed to register and authenticate with the VoIP proxy server.
SIP Proxy Address - the network address of the VoIP proxy server.
SIP Proxy Port - this is the network port the VoIP Phone should use to communicate with the proxy server. Port 5060 is a standard port used in VoIP systems, but may be modified as required.
Display Name - the string used for Caller ID name purposes.
Password - credentials that must be entered.
Registration Expiration - determines the interval in which the VoIP line will attempt to re-register with the Proxy. The proxy may override this setting with a value of its own. If an acknowledgement has not been received from the Proxy within the agreed time, the VoIP card registration information kept in the proxy's database will be cleared. The default registration expiration period is 3600 seconds
SIP Domain Name - SIP domain name to be used if required.
NetBIOS Domain Name - Only editable if Proxy Vendor is set to Skype For Business.
Session Timer - Enables periodic refresh of SIP sessions through a Re-INVITE or UPDATE request. When disabled, the Session Refresher, Session Expiration and Minimum Session Expiration options will be disregarded. If a call unexpectedly disconnects, try disabling this option.
Session Refresh Method - Update or re-INVITE mode can be specified. <PLACEHOLDER>
Session Refresher - in an SIP session that utilizes a session timer, the Session Refresher is the device that will send the periodic Session Refresh requests to refresh the session.

Refresher Options:

Auto - default setting allows both ends of the call to negotiate who will be the refresher. This usually leaves the decision to the device receiving the SIP packets.

UAS - User Agent Server (UAS) is the VoIP device that responds to the SIP Request. For a phone call, it is considered the “called” device. This setting makes sure the SVC-2 card will only negotiate to a Session Timer where the UAS is nominated as the refresher.

UAC - User Agent Client (UAC) is the VoIP device that sends the SIP Request. For a phone call, it is considered the “calling” device. This setting makes sure the SVC-2 card will only negotiate to a Session Timer where the UAC is nominated as the refresher.

Local - this setting makes sure the SVC-2 card will always be the refresher of a Session Refresh.

Peer - this setting makes sure the SVC-2 card will never be the refresher of a Session Refresh.

Session Expiration - determines the interval the VoIP card will try to negotiate with the Proxy to keep the VoIP session alive. Note that the proxy may override this setting with a value of its own. If a Session Refresh request is not properly received by both parties within this agreed time, the session will expire and the call ended. This may be set between 90 to 65535 seconds; the default value is 1800 seconds.
Minimum Session Expiration - if the proxy tries to override the Session Expiration value as specified in the VoIP card, the time entered in this field will be the minimum value allowed. This may be set between 90 to 65535 seconds; the default value is 90.
Transport - designates the connection protocols for SIP data transfers.

Transfer Options:

UDP - User Datagram Protocol (UDP) is "connection-less" and data packets may be sent without negotiation. There is no handshake or setup; packets can be delivered out of order or left out completely. UDP prioritizes speed over accuracy.

TCP - Transmission Control Protocol (TCP) is connection-oriented and a formal connection between endpoints must be established before any data is transmitted. TCP prioritizes accuracy over speed.

TLS - Transport Layer Security (TLS) is used to encrypt SIP traffic and can verify if a device in the SIP exchange is trusted via certificates. See the following article for more information on TLS:

Note that when TLS is selected as the transfer option, a new window will give further options for browsing/uploading a private key or certificate (see below). Be aware that certificates uploaded via the VoIP webpage will not be shown in the SIPS data in the Tesira software interface. Tesira software must configure certificates or the private key filename, and VoIP will retrieve the files from the provisioning server if the server is configured.

See the following article for more information: Using TLS and SRTP in Tesira VoIP Systems

Signaling Port - the signaling Port is used to direct incoming SIP traffic to the correct Line for communications between the VoIP card and the Proxy. The default port for Line 1 is 5060 and the default port for Line 2 is 5062. These settings should be left at this value unless specified by the network administrator
T1 Timer - this timer is used when sending requests over UDP. If the response is not received within this interval, the request is retransmitted. The retransmission interval is doubled after each retransmission.
Prack - guarantees a reliable and ordered delivery of provisional responses in SIP. PRACK Improves network reliability by adding an acknowledgement system to the provisional Responses. Can be set to None, Supported, Required.
Retransmit Timeout - the total amount of time the card will continue to retransmit a UDP packet that has not been responded to.

Real-time Transport Protocol/Secure Real-time Transport Protocol (RTP/SRTP) Settings

RTP and SRTP are network protocols for delivering voice content over networks. SRTP provides encryption and a required message authentication feature. RTP/SRTP functions available from the VoIP management website are shown in the image below:

A description of each field/function is as follows:

Port Start - the first RTP Port used by this line. Must be between 4000 - 65534 and must be one less than the Port End.
Port End - the last RTP port used by this line. Must be between 4001 - 65535 and must be one more than the Port Start.
Static RTP Port - Static Real-time Transport Protocol Port is the Port number used for RTP traffic.
SRTP - Secure Real-time Transport Protocol (SRTP) provides encryption of the RTP audio data. This is available if Transport is set to TCP or TLS in SIP. SRTP may be set to Disabled, Allowed, Preferred or Required.

See the following article for more information on SRTP: Using_TLS_and_SRTP_in_Tesira_VoIP_systems

G.723 Encoding Rate - defines the G.723 bit rate. The options available are 5.3 and 6.3 kbps.
Suppress RTCP On Hold - this parameter determines whether RTCP packets continue to be sent across the trunk for calls that have been placed on hold.

SIPS

If TLS is selected as the transfer option, further options are available to upload certificates:

SIPS Keyword - keyword used for secure SIP on a per-Line basis. Certificate Preference must be set to Keyword.
Certificate Preference - can be set to Fully Verify, Trust, Keyword, or Accept All.
SIPS URI - SIP messages will have the following format when TLS is selected as the Transport and SIPS URI is set to On: sips:abc@biamp.com. If TLS is used but SIPS URI is set to Off, the format would be: sip:abc@biamp.com.

Line 1/Line 2 Setup