[Newlug] August meeting: SIP and IMS

[telecomtom at vedatel.com]

the meeting time-and-place is (are?):

time:  6:30 p.m. august 13 (second monday of the month)
place: Sports Corner Bar & Grill - Upstairs
       500 Grant Street
       De Pere, WI 54115

Be there or beware! -- and please bring a couple of SIP phones if you have
any, like polycom, snom or grandstream, or headphones with mic and ear
plug/speaker. also need a volunteer to fire up wireshark and display
"traffic of interest" by connecting to the overhead display with a USB
cable (someone have a 10' or 20' USB cable? please bring.)

I have some ideas about IMS that border on conspiracy theory. But we also
have lots of facts about SIP to discuss and demonstrate before that.

-- TT


Darrick wrote:
> Eric Cunningham wrote:
>> Is the next meeting August 13th?  Have we settled on the second Monday
>> of every month?  Website still says July...
>
> Is the location set?  It would be smart to make some sort of
> announcement a few days (at least) before the meeting.  Some of us were
> thinking that it was this Monday.
>
> Darrick
>
>>
>>
>> telecomtom at vedatel.com wrote:
>>> meeting fans:
>>>
>>> since no one has volunteered to make a presentation at the august
>>> meeting,
>>> I'll talk about SIP.
>>>
>>> There is a good tutorial slide show at the iptel.org website:
>>> http://www.iptel.org/files/sip_tutorial.pdf
>>> It's too long for the meeting (242 pp.), so I'll pick a few slides
>>> that we
>>> can focus on (like the section on IMS starting on p. 196). If anyone is
>>> interested in other slides, feel free to point them out during the
>>> discussion.
>>>
>>> SIP is the future, and to a large extent the present, of
>>> communications.
>>> it runs over IP: wherever there is IP, SIP call control can be
>>> supported.
>>> you can set up, maintain and tear down any kind of call with SIP:
>>> voice,
>>> video, even tactile-sensory calls (if they ever get the body suits
>>> down to
>>> an economical price). SIP call control usually, but not always, is
>>> associated with RTP/RTCP as the media content. I can go through a few
>>> basic scenarios to show what SIP does to set up a call and how RTP then
>>> handles the media/content flow. When the callers are finished, they use
>>> SIP to disconnect / shut down the call.
>>>
>>> Those are the two most common scenarios: P2P and PBX. Peer-to-peer
>>> calls
>>> can be made without the aid of a PBX like asterisk or sipx. We can do
>>> both
>>> at the meeting if time permits. I hope someone will have a packet
>>> sniffer
>>> (like wireshark) so we can examine SIP control packets and the RTP
>>> content
>>> packets during the demonstation.
>>>
>>> So there you have it:
>>> * SIP control, RTP content flow
>>> * outline a half dozen or so SIP commands
>>> * technical point: RTP vs. RTCP
>>>   RTCP packets are seldom used, but they can be significant
>>> * voice, video, or sensory-in-general calls
>>> * for voice: commonly use G.711 or G.729 as the codec
>>> * tramsmit/send vs. receive:
>>>   the codec for caller A can differ from caller B:
>>>   for example, caller A can transmit using G.711 and caller B
>>>   can transmit with G.729 encoding.
>>> * P2P vs. PBX
>>> * IMS: IP Multimedia Subsystem:
>>>   A little history of the the so-called softswitch and the
>>>   International Packet Switching Consortium that preceded IMS, back
>>>   in the days when some telecom companies were brave enough to try
>>>   to bypass the incumbent telcos.
>>> * and maybe a few other issues
>>>
>>> -- TT
>>>
>>>
>>>
>>>
>
>
> --
> Darrick Hartman
> DJH Solutions, LLC
> http://www.djhsolutions.com
> http://www.djhsolutions.com/wiki
>