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Brekeke SIP Server Wiki

  1. Brekeke SIP Server Basic Info
  2. Security
  3. Update & Upgrade
  4. Configuration
    1. Installation
    2. Basic Setup
    3. BSS Administration Tool
    4. Network Configuration
    5. Dial Plan
      1. Dial Plan Basics
      2. Dial Plan Reference
      3. Dial Plan Tutorial
      4. Dial Plan Examples
        1. Allow Parentheses in Contact-URI
        2. Apply two plugin to one call
        3. Authenticate callers by IP address
        4. Brekeke SIP Server works as Redirect Server
        5. Change Contact header
        6. Change payload type
        7. Change RTP Connection Info
        8. Connect to Gateways with BSS
        9. Count registration records per user
        10. Decline calls from non-numbered callers (SIP ID)
        11. Define outgoing From and To headers with / method
        12. Disable RTP-Relay if SDP indicates a Public IP address
        13. DNS-SRV failover
        14. Execute SQL query from Dialplan
        15. Failover with the Dial Plan
        16. Limit Concurrent Sessions through a specific route
        17. Limit session per sec in Dial Plan
        18. Remove all current registrations
        19. Reject non-registered caller's call
        20. Restrict INVITE Requests
        21. Restrict REGISTER requests
        22. Restrict registration only from single UA
        23. Restrict Upper or Thru Registration
        24. Set time range in Dial Plan
        25. Use Response Header Definition
        26. $webget method examples
      5. Dial Plan FAQ
    6. Block List
    7. Provisioning
    8. Push Notification
    9. SDN
    10. STIR/SHAKEN
    11. Advanced Edition
    12. Other Functions
    13. Third-Party Database
    14. Environment Variables
  5. Troubleshooting
  6. Developer's Guide
  7. Version History

Change RTP Connection Information

With variable “&net.rtp.ifsrc.audio” in Dial Plan rule [Deploy Patterns], Brekeke SIP Server will add the the IP defined in the variable as RTP connection IP address under INVITE SDP audio Media Description.

Rule:
--------------------
[Matching Patterns]
$request = ^INVITE

[Deploy Patterns]
&net.rtp.ifsrc.audio = <IP>
--------------------

Replace <IP> with the IP needed to set for RTP connect information in SDP audio Media Description.
Also, the variable “&net.rtp.ifdst.audio” can define the destination side of RTP connection IP address.

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