FS weekly 2011 05 11

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Contents

Calling Instructions

Wednesday May 11th, 2011 at 1700 UTC (1200 CDT / 1pm EDT)
sip:888@conference.freeswitch.org or via the good old PSTN at +1-919-386-9900 or +44-3300-100-295
gtalk:conf+888@conference.freeswitch.org Or click on this link
Or call Skype the skype user "skypiax5", you'll be automatically connected (max 20 concurrent users).
Codecs: PCMU/PCMA, G.722, CELT, Speex, Skype, among others
Press 0 to mute/unmute your self. Press * to deaf/undeaf.

  • What are the Beeps in the conference? One Beep - someone has joined, two beeps - someone has left.

Agenda

Please add agenda items as needed. If you have a question you'd like to ask then edit the Items Needing Discussion section below. Be sure to put your name/nick on your question.

Note to Collins: RECORD THE CONFERENCE!!!

News, Notes, & Miscellaneous Fun Stuff

  • ClueCon schedule to be posted soon, stay tuned!
  • New sounds version 1.0.16
    • check 'em out
    • Translators, please see about getting the new sound files x-lated
    • Note the new compare.pl script I wrote and put into $fs_src/docs/phrase/

Featured Presentation

  • Community discussion: What do you think about MS + Skype? Does it affect us? Is there opportunity for FS to gain traction by being part of an OSS alternative?
    • Anyone here used Jitsi? (Formerly SIP Communicator)

Upcoming Presentations

  • Eliot Gable's HA stuff (no date yet)
  • Polycom (still working on a date)
  • All Upcoming Dates Open - Please let me (MSC) or NormT know if you have a topic or idea.

Questions For Developers

  • Freeswitch AAA using Radius *
    • cdr is ok, what about AAA, authentication, autorisation/accounting?
  • Database Submodules for mod_db
    • can we embed pgsql library or mysql library, in fs and build mod_db_mysql and mod_db_pgsql?
    • can ODBC be moved to mod_db_odbc?
    • the strategy look like mod_spydermonkey submodules**
  • can all codecs be moved to mod_spandsp like mod_speex, mod_ilbc and be enabled/disabled through spandsp.conf.xml?

Janitorial Items

  • Add your items here

Items Needing Documentation

  • Add your items here

Stuff started but needs some community input

  • Add your items here

Suggestions For Future Meetings & Future To Dos

  • Math: Sofia internals
  • Eliot Gable: mod_ha_cluster
    • What is mod_ha_cluster?
      • In short: N+X (N Masters + X slaves) HA replacement for "Pacemaker + Corosync managed FS"
    • What is planned for mod_ha_cluster?
      • In short: manage interfaces, IPs, firewall rules, master/slave status; eventually, share registrations and call states without shared DB and do 'sofia recover' on failover
    • When will it be available?
      • In short: Master/slave, interface, IPs, and firewall management within the next couple of months (if I have the amount of time available that I expect to have)
    • For additional details, or if you would like to beta test, be on the call

Presenters Needed For These Topics

  • SIP 101 (beginning SIP, how does it work, how to look at different SIP packets)
  • mod_shout/shoutcast, esp with one-way conferences for scalability
  • Codec negotiation - early vs. late, why you need it, how to do it
  • T.38 - what it is, how to use it, etc.
  • Multi-tenancy (bounties welcome)
  • Steve Underwood - SpanDSP, T.38, etc. with FreeSWITCH
  • mod_fifo vs. mod_callcenter - why use one or the other? Strengths and weaknesses of each
  • embedding FS in other applications (libfreeswitch)
  • IPv6 - what it is, how to use, differences with IPv4, how to configure FS

FreeSWITCH High Availability Testing

Weekly FreeSWITCH High Availability testing during the conference call. At the end of the weekly conference call the lead developer of FreeSWITCH (anthm) will intentionally crash and restart the conference server. This tests the high Availability that is built into FreeSWITCH. Your call should continue once the server is restarted.

Depending on what type of NAT device will determine if your call is "Re-Invited" back to the conference.

Need SIP Trunking or DIDs?

If you are looking for SIP Trunking, Origination or Termination please visit www.VoiceNetwork.ca

  • Unlimited Incoming DIDs from $3.95
  • Per Minute DIDs from $0.99 @ 0.01 per minute
  • Outbound Canadian Termination from $0.005 and USA from $ 0.0125

See Also

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