SIP Settings

The SIP configurations require professional knowledge of SIP protocol, incorrect configuration may cause calling issues on the SIP extensions and SIP trunks.

Go to Settings > PBX > General > SIP to configure the SIP settings.

SIP General Settings

Option Description
UDP Port UDP Port used for SIP registrations. The default is 5060.
RTP Port

RTP Port for transmitting data. The From-port should start from 10000. From-port and To-port should have a difference value between 100 and 10000.

The default is 10000-12000.
TCP Port TCP Port used for SIP registrations. The default is 5060.
Local SIP Port A random port in the port range will be used when sending packets to SIP server. The default range is 5062-5082.
Registration Timers
Max Registration Time Maximum duration (in seconds) of incoming registrations and subscriptions. The default is 3600 seconds.
Min Registration Time Minimum duration (in seconds) of incoming registration and subscriptions. The default is 60 seconds.
Qualify Frequency How often to send SIP OPTIONS packet to SIP device to check if the device is up. The default is 30 per second.
Outbound SIP Registrations
Registration Attempts The number of registration attempts before giving up (0 for no limit).
Default Incoming/Outgoing Registration Time

Default duration (in seconds) of incoming/outgoing registration. The default is 120 seconds.

Note: The actual duration needs to minus 10 seconds from the value you filled in.
Subscription Timer
Max Subscription Time Maximum duration (in seconds) of incoming subscriptions. The default is 3600 seconds.
Min Subscription Time Minimum duration (in seconds) of incoming subscriptions. The default is 90 seconds.

NAT Settings

If your PBX is operating in a network connected to the internet through a single router, your PBX is behind NAT.

The NAT device has to be instructed to forward the right inbound packets (from internet) to the PBX server.

Note: You need to configure NAT settings when you want to register a remote extension to the PBX or when you need connect to the PBX via SIP trunk.

Yeastar S-Series VoIP PBX supports 3 methods to configure NAT.

SIP Codec

A codec is a compression or decompression algorithm that used in the transmission of voice packets over a network or the Internet.

Codec Selection
Yeastar S-Series VoIP PBX supports G711 a-law, u-law, GSM, H261, H263, H263P, H264, SPEEX, G722, G726, ADPCM, G729A, MPEG4 and iLBC.
Note:
  • You need to choose at least one same code on the PBX and on your phones, or there may be a problem of the call.
  • If you want to make video calls, you need to select H261, H263, H263P, H264 or MPEG4 codec on the PBX and on your phones.
iLBC Settings
The iLBC codec supports two modes: 20ms and 30ms frame length modes,
To get better voice quality, you need to set the iLBC mode according to your SIP endpoints.
Note: Linkus uses iLBC 20ms mode. When Linkus is enabled, this option is switched to 20ms mode automatically.

TLS Settings

Option Description
Enable TLS Check the checkbox to enable TLS.
TLS Port TLS Port used for SIP registrations. The default is 5061.
Certificate Choose the TLS certificates.
TLS Verify Server If set to no, don't verify the servers certificate when acting as a client. If you don't have the server's CA certificate you can set this and it will connect without requiring TLS CA file. The default is no.
TLS Verify Client If set to yes, verify certificate when acting as server. The default is no.
TLS Client Method Specify protocol for outbound client connections. The default is sslv2.

Session Timer

A periodic refreshing of a SIP session that allows both the user agent and proxy to determine if the SIP session is still active.

Option Description
Session-timers

Choose the session timers mode on the system:

  • No: Do not include “timer” value in any field
  • Supported: Include “timer” value in Supported header
  • Require: Include “timer” value in Require header
  • Forced: Iclude “timer” value in both pportednd equired header.
The default is Supported.
Session-Expires The max refresh interval in seconds.
Min-SE The min refresh interval in seconds, it must not be less than 90.

Qos

QoS (Quality of Service) is a major issue in VoIP implementations. The issue is how to guarantee that packet traffic for a voice or other media connection will not be delayed or dropped due interference from other lower priority traffic.

When the network capacity is insufficient, QoS could provide priority to users by setting the value.

Option Description
ToS SIP Type of Service for SIP packets.
ToS Audio Type of Service for RTP audio packets.
ToS Video Type of Service for RTP video packets.
Cos SIP Class of Service for SIP packets.
Cos Audio Class of Service for RTP audio packets.
Cos Video Class of Service for RTP video packets.

T.38

Adjust T.38 settings if T.38 Fax don't work.

Option Description
No T.38 Attributes in Re-invite SDP If this option is selected, SDP re-invite packet will not contain T.38 attributes.
Error Correction Enable or disable Error Correction for the fax.
T.38 Max BitRate Adjust the max BitRate for T.38 fax.

Advanced SIP Settings

Option Description
Allow RTP Re-invite By default, the system will route media streams from SIP endpoints through itself. Enabling this option causes the system to attempt to negotiate the endpoints to route packets to each other directly, by passing the system. It is not always possible for the system to negotiate endpoint-to-endpoint media routing.
User Agent Change the User-Agent field.
Send Remote Party ID Whether to send Remote-Party-ID in SIP header or not.
Note: This configuration only take effects on internal calls. To set up for external calls, configure the Advance settings of SIP trunk.
Send P Asserted Identify Whether to send P-Asserted-Identify in SIP header or not.
Note: This configuration only take effects on internal calls. To set up for external calls, configure the Advance settings of SIP trunk.
Send Diversion ID Whether to send Diversion in SIP header or not.

If this option is selected, the Diversion value will be extension number.

Note: This configuration only take effects on internal calls. To set up for external calls, configure the Advance settings of SIP trunk.
Support Early Media Whether to support Early Media or not.
All Busy Mode for SIP Forking
  • Check this option: When one of the terminals that register the same extension number is busy in a call, the other terminals will not receive calls.
  • Uncheck this option: When one terminal is busy, the other terminals will still be able to make and receive calls.
Inband Progress This Inband Progress setting applies to all the extensions.
Note: To configure global Inband Progress setting, you need to contact Yeastar support to configure a custom config file.
  • Check this option: PBX will send a 183 Session Progress to the extension when told to indicate ringing and will immediately start sending ringing as audio.
  • Uncheck this option: PBX will send a 180 Ringing to the extension when told to indicate ringing and will NOT send it as audio.
Get Caller ID From Decide the system will retrieve Caller ID from which header field.
Get DID From Decide the system will retrieve DID from which header field.
Note: If Remote-Party-ID is selected but the SIP trunk doesn't support this, the system will retrieve DID fron INVITE header.
100rel Whether to support 100rel or not.
Allow Guest If this option is selected, PBX will accept the unknown calls.
Support Message Request Whether to support SIP Message Request or not.
Maxptime Select or enter the Maxptime value.
DTMF Passthrough

If DTMF Passthrough is enabled, PBX will not process the DMTF tones, and passe DTMF tones transparently to the other end.

Enable uaCSTA connection If this option is enabled, the PBX will use uaCSTA (User Agent Computer Supported Telecommunications Application) to remotely control the IP Phone via Linkus Desktop Client CTI. Your IP Phone needs to support uaCSTA standard to use this function.