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Telium Support Group
Post count: 263

If you are willing to accept the risks of placing new single points of failure in your call path, and you are not using the OEM edition of HAAst (which includes call survival features), then yes you still have options. The key to this solution is to ensure directmedia (RTP flowing directly between endpoints). It’s also quite likely that your endpoints will expect to see the SIP channel responsive as well (or they may drop the call).

Establishing directmedia involves:

  • Ensuring the media anchor points are accessible to one another without NAT.
  • Ensuring Asterisk is configured to use re-invites/directmedia
  • Ensuring your Asterisk dialplan does not force Asterisk to remain in the RTP stream
  • Ensuring your endpoints do not require transcoding (performed by Asterisk)

Optional: ensuring the SIP endpoints continue to see active SIP connections involves:

  • Placing a B2BUA (or gateway/proxy/SBC) between endpoints and the cluster – this device must place itself into the SIP stream and optionally allow NAT traversal
  • Configuring the B2BUA to allow the interior leg of the SIP call to drop, but keep the outer leg of the SIP call to remain active
  • Configuring the B2BUA to use UDP for SIP (at least for cluster facing leg). This is not always required

For example (this shows two B2BUA’s for clarity, but you can adjust to fit your need):
Keeping calls up

There are open source B2BUA products which might be modifiable to do what you want (eg: the SIPpy project available at: https://github.com/sippy/b2bua). Keep in mind that you are creating a free version of the commercial solution we do not recommend. If this is a critical call center you may be better off developing a proper B2BUA from scratch to do what you want, including moving calls through the new active HAAst node, etc but that is a large undertaking.

HAAst OEM edition creates a call anchor on the PBX, so that even if Asterisk fails the calls don’t drop. HAAst will move the calls to the other node in an orderly fashion (move by IP or SIP redirect), or HAAst will grab the calls by force should the entire PBX server fail.

  • This reply was modified 4 years, 3 months ago by WebMaster.
  • This reply was modified 4 years, 3 months ago by WebMaster.