Back to back UserAgent (b2b)
Description
A role that organizes switching between SIP devices by creating two-way dialogs.
Performs routing and calling of relevant devices with media flow organization between them via roles mgc and mg.
Reserved and scaled in mode Active-Active.
Services:
- Organization and maintenance of SIP-dialogs
- Managing the business logic of media context.
- Routing in the internal numbering plan.
- Calculation of subscriber representation.
- Forking (calling multiple registered devices or in multiple directions).
- Subscriber features: call forwarding, group number, intercept, intercom, intrusion, transfer, shoulder swap, multiple call….
- Chief-Secretary type groups.
- CDR Event Generation.
- Authorization of SIP calls.
- Maintenance of registrations.
- Maintenance of subscriptions (BLF, voicemail) and sending notifications.
- SIP service balancer.
- …
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Mode of operation |
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Backup mode |
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Types of sites |
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Placement |
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Saving and restoring state on reboot |
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appendix |
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Parameters
Name | Type | Default | Description |
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required |
Name. May consist of Latin letters and numbers, must begin with a letter. |
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required |
Role Type. Possible values: |
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required |
The alias of the server network interface on which the roles will interact internally with each other. |
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empty |
Additional role options. Contains a json object or list. |
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empty |
Role activity flag. When set to false, the role does not participate in validation and is not started. |
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required |
Role ID. |
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required |
An indication that the role has been allocated to a separate node. |
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required |
Local UDP port for SIP. The same port handles both TCP. |
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required |
Local TCP port for SIP. If a value other than UDP is specified, it will be raised additionally. |
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Switch for logging events to log |
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Media function logging switch |
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Switch for logging the communication protocol with the MGC controller to the logger |
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Switch for logging dialog state machine trace to logs |
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SIP traffic logging switch to log |
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By default, SDP-offer for unencrypted media stream (rtp) is sent to TLS-connected addresses when initiating calls. You can use the parameter to enable SDP-offer initiation with encrypted media stream for TLS addresses. |
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Codec set sending mode switch in responses to INVITE.
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Place statistics on RTP packets in the call termination event |
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Switch for sending re-INVITE to the call initiator immediately after the dialog is established (SIP confirmation request received) ACK). |
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Switches the device verification mode by sending SIP request OPTIONS when a REGISTER request exceeding the license limit is received.
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List of audio codec names used for call initiation (shoulder call). As value - a list containing names of audio codecs in format
The three basic formats, |
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empty |
List of video codec names used to initiate a call (shoulder call). As a value - a list containing video codec names in the following format
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Utilize video transcoding.
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Dialog recording switch with IVR. |
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Conference dialog recording switch. |
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A switch to record eavesdropping and prompter dialogs. |
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empty |
Headline Trolling. Format:
The default is blank. |
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true |
Switch to downgrade the priority of local MG media gateways located on the same server as the currently active signaling server. |
Configuration example
The configuration is managed in an application available to master domain administrators. The application hides the full content of the configuration, but it is nevertheless accessible via the API.
The configuration contains a section to describe all instances of all roles. Parameters are defined for each specific role instance.
{
"name": "b2b1",
"roletype": "b2b",
"iface": "eth0",
"roleid": 11240,
"separate": true,
"udp": 5090,
"tcp": 5090,
"send_reinvite_on_ack": false,
"allow_ac_set": false,
"check_by_options_on_limit": true,
"record_conf": false,
"record_ivr": false,
"log_cdr": true,
"log_trn": true,
"log_sip": true,
"log_media": true,
"log_mgct": true,
"fwd_headers": [
"X-Header-1",
"X-Header-2"
],
"payloads_audio_offer": [
"PCMU/8000", "CN/8000", "G722/8000"
]
}