Service Providers offer SIP trunks that provide flexible and cost-effective WAN solutions for MiVoice Business. SIP trunks allow MiVoice Business to connect to the Service Provider through the SIP protocol over the IP network. (See the diagram below.) On the client side, SIP phones are starting to proliferate across the full spectrum of telephony vendors. The SIP Trunking solution provides basic feature functionality, billing capability, Emergency Services support, FAX support, and more. See the descriptions below for more information.
SIP Trunking supports 9-1-1 emergency service. The SIP Service Provider can be chosen as the outgoing emergency route. Ensure that the CESID information is programmed.
The Service Provider bills calls based on the peer connection to MiVoice Business. SMDR records are created with a special tag entered in the SIP Peer Profile form for outgoing and incoming calls.
Call Billing for SIP Gateway introduces several enhancements to ensure that the correct billing information is sent to the Service Provider. For all SIP trunk calls, MiVoice Business sends the appropriate billing information to the trunking gateway. Usually, the caller's or diverting party's Billing Number (configured in the Associated Directory Numbers or the User and Services Configuration form) is sent as the billable number. However, you have the option to configure a unique Default CPN and Default Billing Number for each zone in a cluster. These numbers will apply to all devices in the zone and can be sent to a gateway to identify the calling or diverting party and to provide the billing number. You also have the option to provide both numbers: the Default Billing Number and the user-specific billing number For details, see Call Billing for SIP Gateway.
Not supported over SIP at this time.
For 3300 ICP Release 7.0, one restriction exists on a SIP-to-DISA call. During the call, the user cannot enter # followed by a remote number.
Communication between SIP Service Providers and MiVoice Business can be configured to use either Fully Qualified Domain Names (FQDN) or IP Addresses. The Network Elements form provides for the configuration of both for the SIP Peer. On the SIP Peer Profile form, the user can program the Local address as either a FQDN or an IP Address.
The SIP Trunking for Service Provider configuration supports FAX calls over G.711. Attempts to switch to T.38 are rejected, and the call continues as G.711. It is recommended that FAX machines be connected locally or through TDM to the MiVoice Business system that is connected to the Service Provider through SIP trunks.
The SIP Satellite Office Solution configuration supports FAX calls over T.38. For more information on this configuration, see SIP Satellite Office. FAX routing can be configured through third party gateways using either prefix routing or COR routing. Prefix routing chooses a route based on a dialed prefix. COR routing chooses a route based on the COR group restriction on a route. If a particular device belongs to a COR group, it may be restricted from dialing out a particular route and then dials out the next route available in the route list.
FAX tone detection is used to disable the adaptive jitter buffer and echo cancellation to improve the reliability of FAX transmissions over IP networks. FAX tones can be detected in one direction (the TDM side) on the following types of calls: IP trunk to TDM, and SIP trunk to TDM. The Class of Service Option "Campon Tone Security" is used to limit the codec selections to G.711 for Fax calls.
Forking is the ability to split or fork a call so that several locations can ring at once. The first location to answer takes the call. MiVoice Business supports up to eight responses to a SIP request to multiple SIP addresses. If a ninth response is received, the entire call is dropped.
MiVoice Business supports forking for outgoing calls by external SIP forking servers. Incoming calls cannot be forked because MiVoice Business does not allow multiple devices to register at the same URI.
For incoming SIP calls that are tagged for Malicious Call, the 3300 records the Media IP address and port used remotely. As well, the SIP signalling information is captured. This information cannot be sent to the SIP Service Provider, but the information is recorded if needed.
NOTE: Malicious Call SMDR records are logged on MiVoice Business. SIP endpoints cannot invoke Malicious Call Trace, but is it recommended that SMDR be enabled for SIP devices and gateways.
This option can be configured in the SIP Peer Profile form and may be necessary for the connection to the Service Providers. It allows periodic packets of audio when a one-way connection is detected to keep the NAT firewall open.
Each SIP Trunk can register with a registrar. The registrar is assigned using the Network Elements form.
The 3300 supports the use of a single registration name to register multiple phone numbers (RFC 6140). If required by the SIP Service Provider's configuration, you can provision the Registration User Name in the SIP Peer Profile form as "bnc" (bulk number contact). With this configuration, MiVoice Business will register the trunks using the single registration method: instead of sending a separate registration for each DID, a single registration message is sent instructing the Service Provider to put all the DIDs associated with this MiVoice Business system in service at the same time.
SIP Trunking for 3300 ICP Release 6.0 and later is based on the following specifications.
RFC1321 - The MD5 Message-Digest Algorithm
RFC2617 - HTTP Authentication draft
RFC2782 - A DNS RR for specifying the location of services (DNS SRV)
All Srv RR received are ordered jointly by priority and weight. For simplicity, the current implementation does not support server load balance information conveyed through the change of weight in SRV RR.
RFC2976 - SIP INFO Method
RFC3261 - SIP Session Initiation Protocol
The 3300 ICP/MiVoice Business operates as a Back-to-Back User Agent.
It supports UDP, TCP, TLS transports.
Both SIP-URIs and TEL-URIs are accepted on incoming calls although only SIP-URI's are used for outbound calls.
The Options method used for link management.
The configuration allows for Outbound Proxy Servers.
Sending REGISTER messages is supported but incoming requests for registration are not.
RFC3262 - Reliability of Provisional Responses in Session Initiation Protocol
RFC3263 - Session Initiation Protocol (SIP): Locating SIP Servers
Does not interpret " regexp" provisioned in NAPTR RR.
RFC3264 - An Offer/Answer model with SIP
RFC3265 - SIP-Specific Event Notification
RFC3311 - SIP UPDATE Method
RFC3323 - A Privacy Mechanism for the Session Initiation Protocol (SIP)
RFC3325 - Private Extensions to the Session Initiation Protocol (SIP) for Asserted Identity within Trusted Networks
Configuration allows for sending of P-Asserted-Identity and use of the Privacy header when privacy is requested.
RFC 3326 The Reason Header Field for the Session Initiation Protocol (SIP)
RFC3515 - The Session Initiation Protocol (SIP) Refer Method
REFER may be received by the 3300 but will not be generated
RFC3725 - Best Current Practices for Third Party Call Control (3pcc) in the Session Initiation Protocol (SIP)
RFC3824 - Using E.164 numbers with the Session Initiation Protocol (SIP)
RFC3891 - The Session Initiation Protocol (SIP) "Replaces" Header
RFC3892 - The Session Initiation Protocol (SIP) Referred-By Mechanism
RFC3966 - The Tel URI for Telephone Numbers
RFC4028 - Session Timers in the Session Initiation Protocol
Partial support, Session Timer is configurable.
RFC4244 - History Info
RFC4566 - Session Description Protocol
Supports G.711 A-law, µ-law, G.729a
RFC4730 - SIP Event Package for Key Press Stimulus (KPML)
RFC4733 - RTP Payload for DTMF Digits, Telephony Tones and Telephony Signals
RFC4904 - Representing Trunk Groups in tel/sip Uniform Resources Identifiers (URIs)
RFC4916 - Connected Identity in the Session Initiation Protocol
RFC5079 Rejecting Anonymous Requests in the Session Initiation Protocol (SIP)
RFC5806 - Diversion Indication in SIP
RFC5876 - Updates to Asserted Identity in the Session Initiation Protocol
RFC6140 - Registration for Multiple Phone Numbers in the Session Initiation Protocol (SIP)
RFC6432 - Carrying Q.850 Codes in Reason Header Fields in SIP (Session Initiation Protocol) Responses
NOTE: Within the 3300 ICP/MiVoice Business, the characters #, *, @, and + have a special use. If these characters are used in a telephone name or number, they may not appear as expected when sent across a SIP Trunk.
Not supported at this time.
The system administrator can select from Always Active, Disabled, or Auto-Detect/Normal for SIP Trunks using the Network Elements form.