Reference12r2:Gateway/Interfaces/SIP: Difference between revisions
(New page: == SIP Registration section == The entry fields for a '''SIP registration''' are: {| |valign=top nowrap=true|'''Name''' |Descriptive name for this registration. |- |valign=top nowrap=true|...) |
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|valign=top nowrap=true|'''STUN Server''' | |valign=top nowrap=true|'''STUN Server''' | ||
|The STUN servers to use. See [[ | |The STUN servers to use. See [[{{NAMESPACE}}:IP4/General/STUN]] for details regarding the format. | ||
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Revision as of 10:48, 23 August 2018
SIP Registration section
The entry fields for a SIP registration are:
Name | Descriptive name for this registration. |
Disable | A switch to temporarily disable this interface without deleting the configuration. |
Type |
|
Transport |
|
AOR | Address of Record: SIP-URI used to register. Enter the registration ID followed by the SIP provider domain name (for example 8111111e0@sipgate.de or 8111111e0@x.x.x.x:5080 if you need to use the IP-address and a different Port number). |
Local Hostname | The Local Domain for SIP Federation enables to select the TLS Certificate according to the Domain Name. On the incoming SIP calls the host part of the URI is removed if equals with the Local Domain configured here, and the user part is used as Name (H323-ID) or Number (E164). |
Local Port | The Local Port that differs from default port 5060 (or 5061 for SIP over TLS) can be configured here for SIP signalling. |
Proxy | DNS name or IP address of the SIP proxy where SIP messages (REGISTER,INVITE,etc) are to be sent to. Proxy can be omitted if domain part of AOR can be used as remote signaling destination. (append ":<port>" if you need a different destination Port) |
STUN Server | The STUN servers to use. See Reference12r2:IP4/General/STUN for details regarding the format. |
Authorization
Username and password for authorization. Username can be omitted if equal to userpart of AOR.
SIP Interop Tweaks
Proposed Registration Interval | Set in seconds, default is 120 seconds |
Accept INVITE's from Anywhere | If disabled, registered interfaces will reject INVITE's not coming from the SIP server with "305 Use Proxy". |
Enforce Sending Complete | Affects handling of "484 Address Incomplete" responses. If enabled and "484 Address Incomplete" is received, the call is cleared. If not enabled and "484 Address Incomplete" is received, the call is retained and re-initiated in case of new dialing digits. |
No Video | Removes Video Capabilities from outgoing media offer. |
To Header when Sending INVITE | Affects only outgoing diverted calls . Called Party: If set we write CDPN into To header of outgoing INVITE (and DGPN into History-Info header). Original Called Party: If set we write the DGPN into To header of outgoing INVITE (and CDPN into Request-URI). |
From Header when Sending INVITE | Controls the local URI (From header) of outgoing calls. Applys to registered interfaces only. Fixed AOR: Fixed AOR is used as From-URI regardless of the actual calling party number. AOR with CGPN as Display: Fixed AOR is used as From-URI and calling party number is added as display string. CGPN in user part of URI: Variable From-URI with actual calling party number. |
Identity Header when Sending INVITE | Controls the identity header (P-Preferred-Identity, P-Asserted-Identity and Remote-Party-Id) sent on outgoing calls
|
Reliability of Provisional Responses | Controls the support of PRACK (RFC-3262). Supported: Supported as optional extension. Required: Required as mandatory extension. Disabled: Hide support for PRACK extension. |
Advanced | Allows the configuration of additional, not further documented, interop tweaks(e.g. /pai on). The same tweaks can be configured also globally(i.e. not for this SIP-Interface) at the SIP(or TSIP/SIPS)-module. Any tweaks configured at the SIP-Interface will overwrite globally configured tweaks. |
More frequently used Advanced Parameters
There are some options which influence the stack behaviour to handle ambiguities in the SIP standard:
- /pai on
- send identity URI in
P-Asserted-Identity
header. By default, it is sent in theP-Asserted-Identity
header for calls from the PBX to the endpoint, in theP-Preferred-Identity
header otherwise - /ppi on
- send identity URI in
P-Preferred-Identity
header
Other options are available which instruct the stack to use non-standard or deprecated behaviour. Note that this should only be used in rare cases. Better have the vendor of your 3rd party SIP equipment fix its stack implementation:
- /send-deprecated-diversion-header on
- send call history information in the deprecated
Diversion
header in addition to theHistory-Info
header - /single-audio-description
- don't send SAVP+AVP, but SAVP or AVP media description in SDP
- /disable-digest-replay-check on
- disables safeguard against replay attacks
- /no-authentication-info
- don't send
Authentication-Info
header in REGISTER response - /take-sendonly-as-inactive on
- some endpoints use sendonly instead of inactive
- /take-zero-addr-for-hold
- some endpoints use a null IP address (such as in
c=IN IP4 0.0.0.0
) as request for hold in SDP - /send-no-historyinfo
- don't send call history information (i.e. call forward) in outgoing Invite
Media Properties
The configuration of the media properties is evaluated for calls from/to this interface to/from a physical (ISDN, analog, TEST, ...) only. If media relay is active for a call using this interface an 'exclusive' coder config is used to prohibit the use of any other coder. This 'exclusive code media-relay' config can be used to solve interop problems with other equipment which does not support media renegotiation, because with this config no media renegotiation will be performed.
For more information see Media Properties