SIP Settings
This topic describes the SIP settings on the Yeastar P-Series Cloud Edition 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
to configure SIP settings.SIP general settings
Setting | Description |
---|---|
Basic Settings | |
SIP UDP Port | UDP Port used for SIP registration. The default value is 5060. |
SIP TCP Port | TCP Port used for SIP registration. The default value is 5060. |
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.
Setting | Description |
---|---|
iLBC Mode | The iLBC codec supports the following modes: 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,
Opus. Note:
|
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 Client | |
TLS Connection Method | Specify a protocol for outbound client connections.
Note: It's recommended to use the more
secure TLS V1.2.
|
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.
|
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.
|
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 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 | |
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.
|
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.
|