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Saturday, January 9, 2010

Best of New in Kamailio 3.0.0 - #15: sctp

SCTP - Stream Control Transmission Protocol - read more about this protocol at wikipedia or the IETF RFC. Being designed with telephony in mind, SCTP and SIP match perfectly, unfortunately not many end devices support it due to late appearance, but for internal infrastructure and peering is the best option in terms of server resource usage, scalability and high availability.

Kamailio (OpenSER) 3.0.0 offers of the most complete SCTP implementation in the open source SIP and VoIP world. The numbers of features talk themselves:
  • multi-homing
  • statistics for SCTP
  • connections associations - multi-streaming
  • connection auto-close
  • runtime re-configuration of SCTP parameters
  • management of attributes for retransmission at SCTP transport layer
  • connection reuse
  • connection tracking
  • black-listing at SCTP level
  • ability to fallback to SCTP for big UDP packets
  • over 20 parameters to tune SCTP behaviour
  • configurable time to live, send retries, send and receive buffer size, retransmission timeout, a.s.o
  • admin access via RCP control interface
Next some examples of using command line interface application sercmd to access SCTP options:

 $sercmd core.sctp_info
{
opened_connections: 0
tracked_connections: 0
total_connections: 50007
}

$ sercmd core.sctp_options
{
sctp_socket_rcvbuf: 54784
sctp_socket_sndbuf: 54784
sctp_autoclose: 180
sctp_send_ttl: 32000
sctp_send_retries: 0
sctp_srto_initial: 3000
sctp_srto_max: 60000
sctp_srto_min: 1000
sctp_asocmaxrxt: 10
sctp_init_max_attempts: 8
sctp_init_max_timeo: 60000
sctp_hbinterval: 30000
sctp_pathmaxrxt: 5
sctp_sack_delay: 200
sctp_sack_freq: 0
sctp_max_burst: 4
}
Integration of SIP with SS7 networks should be pushed forward by availability of this SCTP implementation in the Open Source space. Also I expect more and more big SIP platforms to migrate internally to SCTP, mainly because of reliability and high availability features offered by this transport layer.

The series will continue with SIP identity authentication - RFC4474

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