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There's no way to perform an unauthenticated SIP INVITE on PrivateServer! You can have different authentication models but you cannot choose to enable an unauthenticated SIP Trunk. |
The two configurations are parallel meaning that the Authentication method is not related with the encryption of the Trunk itself.
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In
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There are two types of trunk that you can createhave two ways, depending on the authentication type :
- username and password: you must insert username and password
- IP based: leave the username and password fields blank and fill in the HOST parameter.
The fields have to be configured as follows:
- NAME: a meaningful name for this trunk
- HOST: IP address of the SIP server provided by ITSP
- OUTBOUND PROXY: IP address of the outbound proxy server provided by ITSP
- VIRTUAL PHONE NUMBER: associated virtual number
- USERNAME: login username
- PASSWORD: password for the username entered above
- PORT: registration port of the service
- REGISTER: select if you want to have only outgoing calls or incoming (see below for deeper explanation)
- CONCURRENT CALLS: maximum number of calls on a single trunk
- NAT: enable/disable the Network Address Translation configuration
- DIRECTMEDIA: experimental sound management in P2P mode without routing the audio stream through the server
- SENDRPID: the Remote Party ID, used to interconnect with VoIP Provider for the management of privacy
- DTMFMODE: tone signaling technology used for this SIP interconnection
- ALLOW: voice codecs used in this trunk
- DISALLOW: voice codecs denied in this trunk
Register option
If the trunk is going to be also used for OUTBOUND calls (external calls) you need to be sure that the "Register" checkbox is checked. In this case the "Name" and "Secret" fields are mandatory as well. Otherwise you can uncheck the "Register" checkbox so the above fields will become optional.
Fill out the fields accordingly to the guidelines above and when you are done click on the "Update" icon at the bottom of the page.
You can get back to the "Sip Trunk List" page where the Sip Trunk table is now filled based on your settings (
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A notification will inform you about the progress of the operation. The number shown in the notification is an internal identifier in the database and thus can be ignored.
To check the entered data of the SIP Trunk click on the "Host" value into the table (for example in this case you should click on "10.0.0.101"). This will open again the Edit Sip Trunk page with the previously entered data.exposed by your peer PBX:
- SIP Account
- IP based
4.3.1 SIP Account authentication
If your PBX peer on the other end of the Trunk has given to you a SIP Account to be used for the Authentication, then you need to insert login and password in Username and Password fields. Plus you should enable the Register option (check
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Typically this configuration is used by public SIP providers because they can't rely just on the IP authentication (the customer's PBX could be behind an ADSL, for instance) and they need to be sure you're just the one who has the right to be connected despite the IP address you are using.
4.3.2 IP Address Authentication
In an enterprise scenarios the authentication is generally based on IP address, because inside enterprise's infrastructure all the structural elements use to have both private and static IP addresses.
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In this case the IP address is most likely an internal one and it should be fixed: these features make it a reliable source for the authentication.
The IP Address Authentication is based on the Host field in the form of the Trunk creation.
Plus you need to enable the Send keep alive option (please see
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The downside of this option is that there will be some more traffic on the socket (each passage of the request is 1.8 KiloByte, thus you can count almost 3.6 KB of traffic every 3 minutes) |
The actual default value for the keep-alive interval is 60 seconds. You can configure the general keep-alive timeout in the NAT configuration form. Please read PSAM 2.4 Advanced configurations to get informations about it.
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Note that the keep alive option can be safely set up even on the SIP account authenticated trunk where it's optional. Instead it considered mandatory for an IP Address Authentication.info |
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