SIP Settings

This topic describes the SIP settings on the Yeastar P-Series PBX System for reference.

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

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

SIP general settings

Table 1.
Setting Description
Basic Settings
SIP UDP Port UDP Port used for SIP registration. The default value is 5060.
Note: If you change the port, the extensions that use UDP protocol must re-register to the new port.
SIP TCP Port TCP Port used for SIP registration. The default value is 5060.
To change the port, select the checkbox of SIP TCP Port and set the port.
Note: If you change the port, the extensions that use TCP protocol must re-register to the new port.
RTP Port Range

RTP port for transmitting data. The default range is 10000-12000.

Note:
  • The From-port value should be greater than 10000.
  • The From-port and the To-port should have a difference value between 100 and 10000.
Outbound SIP Port Range

To prevent from being blocked by carrier due to overloaded calls and subscriptions, you can specify an outbound SIP port range. PBX will select a port from the range to register to the carrier. The default range is 5062-5082.

To change the port, select the checkbox of Outbound SIP Port Range and set the port.

SIP Endpoint Registration Timer
Max Registration Time (s) Maximum duration (in seconds) of incoming registrations and subscriptions.
Min Registration Time (s) Minimum duration (in seconds) of incoming registrations and subscriptions.
Qualify Frequency (s) How often to send SIP OPTIONS packet to SIP device to check if the device is up.
Outbound SIP Registration Timer
Registration Attempts The number of registration attempts before giving up (0 indicates no limit).
Default Registration Time(s)

Default registration duration (in seconds).

Note: The actual duration needs to subtract 10 seconds from the value you fill in.
SIP Endpoint Subscription Timer
Max Subscription Time(s) Maximum duration (in seconds) of incoming subscriptions.
Min Subscription Time(s) Minimum duration (in seconds) of incoming subscriptions.

SIP codec

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

Table 2.
Setting Description
iLBC Mode
The iLBC codec supports the following modes:
  • 20 ms
  • 30 ms
To get better voice quality, you need to set the iLBC mode according to your SIP endpoints.
Codec Selection Select the codec.
Available values: u-law, a-law, GSM, H264, VP8, H263, H263P, iLBC, G722, G726, SPEEX, ADPCM, G729A, MPEG4.
Note:
  • To ensure that users can have audio calls on Linkus Web Client, you must enable at least any one of u-law, a-law, or G722.
  • To ensure that users can have video calls on Linkus Web Client after you subscribe Yeastar P-Series Ultimate Plan, you must enable either VP8 or H264.We recommend that you enable VP8 or set VP8 to a higher priority.

TLS settings

Setting Description
TLS Enable or disable TLS.
SIP TLS Port TLS port used for SIP registration. The default value is 5061.
When PBX acting as a Sever
TLS Certificate Upload a server certificate when PBX acts as a server.
TLS Verify Client Verify client certificate when PBX acts as a server.
Note: If enabled, you need to upload a client certificate to the PBX and TLS client.
When PBX acting as a Client
TLS Connection Method Specify a protocol for outbound client connections.
  • TLS V1.0
  • TLS V1.2
TLS Verify Server Verify server certificate when PBX acts as a client.
Note: If enabled, you need to upload a server certificate to the PBX.

Session Timer

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

Setting Description
Session Timer

Select a session timer mode.

  • No: Do not include “timer” value in any field.
  • Supported: Include “timer” value in Supported header.
  • Required: Include “timer” value in Required header.
  • Forced: Include “timer” value in both Supported and Required header.
Session-Expires (s) The max refresh interval in seconds.
Min-SE (s) The min refresh interval in seconds. The value should not be smaller than 90.

QoS

Quality of Service (QoS) 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 to interference from other traffic of lower priority.

When the network capacity is insufficient, QoS can provide users with priority by setting the value.

Setting Description
ToS (Type of Service)
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 (Class of Service)
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 doesn't work.

Setting Description
T.38 Max BitRate Adjust the max BitRate for T.38 fax.
No T.38 Attributes in re-INVITE SDP If enabled, SDP re-invite packet will not contain T.38 attributes.
Error Correction Mode Enable or disable Error Correction for the fax.

Advanced SIP settings

Setting Description
Incoming Caller ID/DID Retrieval
Get Caller ID From Decide the system will retrieve Caller ID from which header field.
  • From
  • Contact
  • Remote-Party-ID
  • P-Asserted-Identity
  • P-Preferred-Identity
Get DID From Decide the system will retrieve DID from which header field.
  • To
  • Invite
  • Diversion
  • Remote-Party-ID
  • P-Asserted-Identity
  • P-Preferred-Identity
  • P-Called-Party-ID
Note: If Remote-Party-ID is selected but the SIP trunk doesn't support this, the system will retrieve DID from Invite header.
SIP Request Header
User Agent Set the user agent that will be included when sending SIP packages out.
Internal Alert Info

Set an "alert info text" to add to Alert-info header in INVITE request for internal calls.

When receiving an internal call, the phone will inspect "Alert-Info" header to determine which ring tone it should use for ringing.

Other Options
Allow Guest If enabled, PBX will accept unknown calls.
Support Message Request Whether to support SIP Message Request or not.
Inband Progress Whether to enable inband progress or not. The 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 configuration file.
  • Check this option: PBX will send a 183 Session Progress to the extension when told to indicate ringing and immediately start sending ringing as audio.
  • Uncheck this option: PBX will send a 180 Ringing to the extension when told to indicate ringing, but will NOT send it as audio.
Enable uaCSTA Connection If this option is enabled, the PBX will allow user agent Computer Supported Telecommunications Application (uaCSTA) to remotely control the IP phone via Linkus Web Client CTI or Linkus Desktop Client CTI.
Note: Your IP phone should support uaCSTA standard to use this function.