FS weekly 2010 11 10
Wednesday November 10th at 1700 UTC (1200 CST / 1pm EST)
sip:firstname.lastname@example.org or via the good old PSTN at +1-919-386-9900
gtalk:email@example.com 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.
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.
News, Notes, & Miscellaneous Fun Stuff
- OSTAG update
- Chad Philips and Kristian Kielhofner will be presenting a new project
- All Open - Please let me (MSC) or NormT know if you have a topic or idea.
Questions For Developers
- From Chad: what's the difference between "originate_timeout" and "call_timeout"?
Items Needing Discussion
- ClueCon videos update - Mishehu working still, first video available on viddler; see cluecon.com/schedule, Tony's talk description is a clicky link
- Followup from last week: DRK has new info/files to share
- Load balancing - are you doing it? If so, how? Looking for general discussion on what people are doing: SIP proxy on front end, db server(s) on back end, etc. I want to get some information on the wiki so that people have some ideas on where to go to learn more.
- We need someone with some Atlassian skills to assist with our Fisheye setup. It constantly bombs. We may have an old version. In any case, if you know Fisheye and Atlassian please see me or Raymond ([intra]lanman)
Items Needing Documentation
- Add items here
Stuff started but needs some community input
- Add items here
Stuff yet to be documented
- New mod_sofia chan var: sip_force_audio_fmtp (commit msg)
- New FIFO param: taking_calls (commit msg)
- New FIFO flag: no_media (commit msg)
- New FIFO chan vars: fifo_bridged, fifo_manual_bridged (commit msg)
- New stats (in CDRs) rtcp_packet_count and rtcp_octet_count (commit msg)
- sofia_dig API has been in code for some time now, but is still undocumented. (not anymore: sofia_dig)
User Tips & Tricks
If you have something you'd like to share with the community then by all means add it here and we will give you a few minutes on the conference call to discuss it!
LRN - Are you ready?
Are you getting the correct rate on your calls?
Suggestions For Future Meetings & Future To Dos
- Math: Sofia internals
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)
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