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SIP Channels

SIP channels (SIP trunks) are SIP protocol-based communication paths for transmitting calls over IP. They serve as inbound/outbound communication channels with voice service provider PBXes, VoIP gateways or VoIP phones.

SIP channels overview

Creating and managing SIP channels

  • New: Click “New record” in the toolbar
  • Edit: Click the “Edit” button or right-click on the channel
  • Delete: Click the delete button or use the context menu

SIP channel menu

Basic configuration

SIP channel main properties

  • Name: Channel identifier
  • SIP login name: Authorization username
  • Password: Authorization credential
  • Host: IP address, FQDN, or empty (for registration)
  • Outbound proxy: Optional proxy server address in full SIP URI format
  • Domain alias: Alternative domain
  • From domain / From username: Information for SIP headers
  • NAT: Ignores IP/port in SIP headers; communicates via incoming packets
  • ICE support: NAT support for compatible clients
  • Transport: SIP communication protocol (automatically set if empty)
  • Registration: Enable registration to remote PBX/SIP device
  • Registration expiry: Renewal interval
  • Tariff: Billing table selection

Advanced settings

  • Prompt language: Default system prompt language
  • Country tones: Signaling tone configuration
  • DTMF mode: RFC4733 recommended
  • Inband progress: Signal generation setting
  • Qualify frequency: OPTIONS message interval (seconds)
  • Keep-alive packets: Connection maintenance interval (seconds)

Restrictions

  • Max registered contacts: Concurrent client limit
  • Allowed IP addresses: Whitelist (format: 192.168.0.5/255.255.255.255)
  • Blocked IP addresses: Blacklist (same format)
  • Call limit: Maximum concurrent calls
  • CallerID: Custom number for outgoing calls
  • CallerID blocking: Identification presentation options
  • SIP Diversion header: Automatic generation based on RDNIS
  • RPID/PAI headers: Include Remote-Party-ID or P-Asserted-Identity
  • Call transfer: Enable SIP transfer

Audio and video codecs

Codec configuration

  • Audio codec priority: Minimum recommended G.711a (PCMA/ALAW)
  • Video codec priority: Video encoding preference

Extended settings

Extended settings

Registration

  • Max retry attempts: Number of registration retries
  • Retry interval: Time between attempts
  • 403 response interval: Special timing for Forbidden responses
  • Fatal error interval: Timing for 4xx/5xx/6xx responses
  • Permanent authorization rejection: Mark failed attempts as permanent
  • Remove existing contact: Replace previous registration
  • Remove unreachable contact: Automatic removal of unresponsive endpoints

Channel configuration

  • Qualify with authorization: Require authentication for qualify messages
  • Qualify: Qualification based on response (0 = disabled)
  • Overlap dialing: Enable progressive number completion
  • Allow subscribe: Enable subscribe notifications
  • Music class: Music on hold selection
  • COLP method: Connected party update method (Invite/Reinvite/Update)
  • Identification method: Username, IP, auth name or SIP header
  • Inbound/outbound script: Script execution on calls
  • Channel variables: Custom SIP variables

RTP and audio

  • Preferred codec only: Send only the preferred codec in SDP
  • Voice encryption: None / SDES / DTLS
  • Direct RTP: Enable peer-to-peer RTP
  • RTP timeout: Call termination threshold (seconds)
  • RTP hold timeout: Hold timeout (must exceed RTP timeout)
  • RTP keep-alive: Activity generation for NAT traversal
  • Symmetric RTP: Enable bidirectional RTP

FAX

  • Fax detection: Detection and redirect to extension “f”
  • Detection timeout: Time limit for fax identification
  • T.38 support: Enable T.38 protocol
  • T.38 NAT / IPv6: T.38 support over NAT and IPv6
  • T.38 error correction: FEC or redundancy
  • T.38 max datagram: Maximum transfer size (0 = automatic)

Expert settings

Expert settings

Channel configuration

  • 100rel support: Yes / No / Required
  • Min subscribe expiry
  • MWI aggregation: Message waiting indication notification consolidation
  • Redirect method: User / URI core / URI PJSIP
  • REFER: Transfer detail reporting
  • Session timers: Yes / No / Required / Always / Forced
  • Min/Max session timer

RTP and media

  • Optimistic media encryption: Use encryption when possible
  • SRTP tag 32: Use 32-byte tag instead of 80-byte
  • Encryption method: TLS 1.0/1.1/1.2/1.3, SSL v2/v3
  • DTLS verification: Fingerprint / Certificate / None
  • DTLS certificate / private key / CA file
  • DTLS setup: None / Active / Passive / Both
  • DTLS fingerprint: SHA-256 or SHA-1
  • Max audio/video streams
  • RTCP MUX: Combine RTP/RTCP on a single port
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