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.|
|AOR||Address of Record: SIP-URI used to register. Enter the registration ID followed by the SIP provider domain name (for example email@example.com or firstname.lastname@example.org: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 source port for SIP signalling can be configured here. If its empty a random port will be used. Dont use the same static port on multiple SIP Accounts!|
|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 STUN for details regarding the format.|
Username and password for authorization. Username can be omitted if equal to userpart of AOR.
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.
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 .
|From Header when Sending INVITE||Controls the local URI (From header) of outgoing calls. Applys to registered interfaces only.
|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).
|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-Identityheader. By default, it is sent in the
P-Asserted-Identityheader for calls from the PBX to the endpoint, in the
- /ppi on
- send identity URI in
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
Diversionheader in addition to the
- don't send SAVP+AVP, but SAVP or AVP media description in SDP
- /disable-digest-replay-check on
- disables safeguard against replay attacks
- don't send
Authentication-Infoheader in REGISTER response
- /take-sendonly-as-inactive on
- some endpoints use sendonly instead of inactive
- some endpoints use a null IP address (such as in
c=IN IP4 0.0.0.0) as request for hold in SDP
- don't send call history information (i.e. call forward) in outgoing Invite
Disable Interworking of Hold Notifications to SIP Provider
During the tests we concluded that when interworking the hold-notify message to SIP and sending to the SIP Provider two consecutive Re-Invites with "send-only" attributes, the IMS platform replies to the second re-invite with "inactive". By doing so this call is put on hold without any Music on Hold - just silence. To avoid this behaviour we need to disable the interworking of the hold-notify message by this setting:
!config add SIP /no-hr-notify (Alternative: TSIP / SIPS) !config write !config activate
SIP Options Interval (Optional)
Some Provider uses SIP Options to monitor the SIP Trunks, so it's mandatory that Innovaphone replies to incoming SIP Options received. This is done by default. Additionally we can also send SIP Options to the SIP Proxy and have similar mechanism for redundancy. If the remote Proxy doesn't reply to outgoing SIP Options, the Innovaphone Gateway will send the call to the next interface. To enable sending of Options - messages, the following setting must be done:
!config add SIP /options-interval 30 (Alternative: TSIP / SIPS) !config write !config activate
This option will only take effect on connections "without registration". In connections with a registration there already exist a keep alive.
Remove Comfort Noise (CN) Capability from SDP
During the tests we found out that some specific 3rd party devices connected to the IMS network support only a single coder/payload in the offer. When doing the coder negotiation, this devices repeat the coder negotiation until they have only 1 coder in the offer or until they reach a specific number of retries. Since Innovaphone by default always include the payload 13 (Comfort Noise) in addition to the used voice coder/payload, this would make the remote device to do multiple re-invites to try to reach the single coder/payload in the offer. To avoid unnecessary signalling, we should disable the sending of Comfort Noise capability.
!config add SIP /rem-cn-capability (Alternative: TSIP / SIPS) !config write !config activate
Setting of P-Asserted ID instead of P-Preferred ID
When using the feature of ReRouting the call (SIP 302 Move Temporary) to the SIP Trunk, the IMS platform checks the P-Asserted ID setting. By default we send as P-Preferred ID instead, so that will not work. As a result, we need to configure the following setting:
!config add SIP /pai (Alternative: TSIP / SIPS) !config write !config activate