4.0.1 Background on SIP Trunks
The SIP Trunks are used for connecting PrivateServer to a local PBX or to a SIP Voice VoIP provider. The above statement implies that a A Trunk can be used both for receiving and placing calls in a bidirectional way.
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| By Secure Call design the SIP Trunks are available only for the trunks are mainly useful to interconnect EVSS ecosystem with other SIP infrastructure, using an end-to-site security model, featured by PrivateGSM Enterprise Edition. SIP trunks have a very limited usefulness for PrivateGSM Professional Edition can't place calls on a Trunk. |
The PrivateServer is designed to perform always as a client when configuring a SIP Trunk. This means that PrivateServer itself can not act as a Trunk Provider.
Even if each| . |
Each SIP Trunk can be used to both receive and place calls, PrivateServer has to be properly configured in order to act in a bidirectional way. By default the Inbound Trunks are configured for enabling the calls directed to external number through a different PBX. They can be later set up to let the PrivateServer receive incoming calls from a different PBX. This setup is made promoting one Inbound Trunk as an Outbound one.
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| Basically the Trunk setup has to be done for the outgoing calls first and after this function is enabled you can setup it for incoming calls as well. |
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4.0.2 SIP Trunk's
Familiescategories
From the PrivateServer version 2.5 you have two main SIP Trunk's Familiescategories:
- Secure Trunks
- Insecure Trunks
The former ones entail both TLS and SRTP, while the latter none of them: they use just RTP over UDP (thus without TLS). Which one is to be used depends from on many factors, major one is the kind of PBX on the trunk's end. For example Cisco Unity Call Manager and Avaya PBX Unified Communications Manager are usually connected to PrivateServer via Secure Trunk, while SIP providers like Messagenet providers most often use the Unsecure Trunk.
24.0.3 Authentication Models in SIP
TtrunksTrunks
Apart the Trunkfor trunk's Families categories, you should familiarise get acquainted with the Authentication Model, that can be:
- SIP Account for Registered Trunks
- IP Authenticated forUnregistered Trunks
The SIP Trunks using the Registered Authentication Model need an account on the other end. Then they use the account's credential to authenticate the Trunk when the PrivateServer is started and every communication that passes over that Trunk is considered authenticated, thus valid by default, and has the right to be processed and routed.
The SIP Trunks with no registration, on the other hand, don't have needs do not need for an account to be setup, : instead they use the IP address to authorise authorize every SIP communication directed to the PrivateServer. This model implies that any new SIP communication is authenticated against the sender's IP address (meaning the one belonging to the PBX connected to the PrivateServer).
The latter model is mostly used in production enterprise systems, eg with CUCM and Avaya. The former is mostly used with SIP ProviderProviders.
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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. |
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4.0.4 General starting point for SIP Trunk configuration
No matter what kind of Trunk you're going to configure on PrivateServer, you would anyway come throughout the SIP Trunk configuration page in order to complete your setup.
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The fields have the following meanings:
- NAME: a meaningful name for this trunk
- HOST: IP address or Hostname 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
- REGISTER: select if you want to have only outgoing calls or incoming (see below for deeper explanation)
- SEND KEEP ALIVE: select it if you want PrivateServer to send a SIP UPDATE OPTIONS packet on a regular basis
- PORT: registration port of the service
- TRANSPORT: protocol to be used as a transport mean. Choices are:
- UDP
- TLS
- ENCRYPTION: whether the trunk has to be encrypted or not
- MAX 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
- AUDIO TONES: this is used for Audio Messaging. The possible values are:
- NO AUDIO TONES: no audio messages would be played on this trunk
- ON EARLY MEDIA: the audio messages would be played using SIP early offer
- ON ANSWERED CALLS: the audio messages would be played using SIP delayed offer
- DTMFMODE: tone signaling technology used for this SIP interconnection
- ALLOW: voice codecs used in this trunk
- DISALLOW: voice codecs denied in this trunk
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- Inbound
- Outbound
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For creating a new SIP Trunk you go on the Inbound menu voice and then use the New Sip Trunk in the Sip Trunk List page (see
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For each of the Families and the Authentication Models category and authentication models you are going to find a specific manual page describing the correct configuration and the possible parameters to be used.
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