[Cu-wireless] VoIP overview

Sascha Meinrath sascha at ucimc.org
Sat Feb 14 19:08:37 CST 2004

If we were to go with an SIP implementation, what would the next steps be?
Does the work we'll be doing on the OSI-portion of CWN development cover
some of the areas that we'd need to work on?  Are there places where we
could use the OSI stuff as a stepping stone for future VoIP developement?
I'm just trying to get an idea for what would be required to make this
happen.  Once a system of this sort is created, it will fundamentally
change the way phone and Internet systems interoperate, and this presents
an amazing opportunity for our work to have meaningful real-world impacts.


On Sat, 14 Feb 2004, David Young wrote:

> I had a meeting the other day with Stephane Alnet about Voice over IP,
> especially as it concerns wireless. Here is my summary. All mistakes
> are mine.
> Stephane explained that in any VoIP solution, there are two types
> of traffic, "call signaling" (aka "call control") and the "bearer"
> traffic. Call signaling is used to start/stop calls and signal conditions
> like "line is busy."  Bearers carry voice/fax/modem/data traffic.
> There are two major standards for VoIP. SIP is IETF's standard. It
> is pretty new, with commercial implementations 2 years old or younger.
> H.323 is a 10-year old standard by ITU. It comes in versions 1 through 4,
> with 4 very much preferable. It is older, with commercial implementations
> being around for 10 years.
> One reason SIP is expected to prevail is its
> featurefulness/buzzword-compliance.  It supports for "presence," i.e., the
> notion that you might or might not have your cell phone on your person,
> or you might want to take some class of calls at home, others at home
> and on your cell, and so on. One "class" of calls might be those placed
> by your employer, and another class, those placed by your family. (I am
> guessing that there are classifications based on time and such, too.)
> SIP has features for dealing with presence.
> SIP encompasses call signaling and, using a protocol called SDP,
> bearer-traffic negotiation. (Bearer-traffic negotiation is where the
> endpoints of a VoIP call choose a codec and set other parameters for
> the bearer channel.) In H.323, however, there are different protocols
> for signal & bearer traffic. H.225 is the signaling protocol, and H.245
> is the bearer-traffic negotiation protocol. Both SIP and H.323 bearer
> traffic is carried by the Real-Time Protocol (RTP). RTP is an unreliable
> datagram service for data that needs timely, constant-speed delivery
> above all else.  RTP packets are embedded in UDP packets. For every RTP
> stream in one direction, there is usually a Real-Time Control Protocol
> (RTCP) stream in the opposite direction.  RTCP carries statistics on
> percentage & rate of delivery back to the RTP source.
> A SIP phone converts telephone numbers to (IP number, UDP port) pairs
> using a "SIP proxy" device. An H.323 does the same using a "gatekeeper"
> device which speaks a protocol called RAS.
> The community wireless network should probably use SIP.
> There are devices by Cisco and others for connecting a VoIP network to
> the Public Switched Telephone Network (PSTN), aka "the Bell System".
> One device, ATA-186, acts as a VoIP gateway to your FXO (Foreign eXchange
> O...: fancy name for your two-wire phone outlet at home). The telco's
> end of the pair is called the FXS, Foreign eXchange Startloop. This is
> an analog connection to the PSTN. It's cheap, you've got one already.
> You can make a digital connection to the PSTN using ISDN. There are a
> few kinds of ISDN service. One is BRI, Basic Rate Interface. Provides
> two bearers (64kb/s channels, where kb = 1000 bits, not 1024 bits, in
> telco-speak) and a control channel. It is not cost-effective to use
> a BRI as your gateway to the PSTN.  Paying for up to 8 FXOs is more
> cost-effective than buying a digital connection, actually. There is an
> ISDN connection called a PRI, also expensive, that has 23 64kb/s bearers.
> A T1CAS (E1/R2 is the equivalent in Europe) which also has 23 bearers,
> is most cost-effective.
> You can "roll your own" VoIP gateway using a modem.  There is
> an open-source project, Bayonne, that is concerned with making PBXs
> (business phone systems). They use some device that can "bond" eight FXOs.
> There are a variety of codecs, which are essentially encodings for
> voice. G.711 is the PSTN codec. It's not too fancy, works just fine
> on a 64kb/s channel. There is also G.729, a 8kb/s codec, uses voice
> recognition principles to compactly encode whole phonemes (e.g., the
> "ay"-sound). An 8kHz sampling rate is standard for telephony codecs.
> (Incidentally, G.729 traffic is 28kb/s on IP over Ethernet. Lots of
> overhead, there!)
> There is an art and science to figuring out how many trunk lines you
> need to serve some number of households.
> The routers and other equipment on your IP network need to provide good
> quality of service (QoS) to the RTP packets that carry your IP phone
> calls, or else your users will be frustrated, both as they place calls
> and especially during conversations. Typically a router will put the RTP
> packets onto a high-priority or low-latency queue. 802.11 networks have
> their own special problems with quality of service. 802.11e is a task
> group concerned with QoS extensions to 802.11, but they have not issued
> a final standard. (I think WECA---Wireless Ethernet Compatibility
> Alliance---has produced a standard called Wireless Multimedia
> Extensions---WME---which is a subset of the anticipated 802.11e standard.)
> VoIP handsets are plug & play. They will automatically tftp a
> configuration file from the VoIP gateway---which they discover using
> DHCP? To "VoIP-enable" the community wireless network, it looks like it
> may be necessary for each node to be able to bootstrap popular VoIP phones
> and also to act as a SIP proxy. For truly "ad hoc" telephony, it will
> be necessary for the nodes to provide a "distributed telephone book".
> For PSTN-connected networks, we can probably rely on the PSTN gateway
> to provide the phonebook centrally.
> Dave

Sascha Meinrath
Project Manager & President	 *	Project Manager
Acorn Active Media Foundation	***	Eggplant Active Media
www.acornactivemedia.com	 *	www.eggplantmedia.com

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