SIP Technical Overview

Archived from groups: comp.dcom.voice-over-ip (More info?)

Mike Schumann wrote:
> I am in the process of deploying VOIP at a number of office locations. My
> initial efforts have been somewhat disappointing, due to voice quality
> problems.
[...]

Look at what codecs you are using, what the link delay and jitter are,
and work from that.

You want minimum delay and jitter. Sometimes the lower bandwidth codecs
help in that you can run a link at lower than max capacity to improve
the link performance.

Don't share VoIP and internet traffic through the same bottleneck!!!

.... And then you can play with VLANs and QoS...

Good luck,
Martin

--
---------- OS? What's that?!
- Martin - To most people, "Operating System" is unknown & strange.
- 53N 1W - Mandrake 10.0.1 GNU Linux - An OS for Supercomputers & PCs
---------- http://www.mandrakelinux.com/en-gb/concept.php3

Archived from groups: comp.dcom.voice-over-ip (More info?)

Why hasn't anyone mentioned www.voip-info.org?
It has EVERYTHING to do with VoIP, an excellent resource

-Paul

"DJC" <email@iqws.com> wrote in message
news:42025b88$0$19164$cc9e4d1f@news-text.dial.pipex.com...
> i have found the following site is very useful:
>
>
> http://www.vovida.org/
>
>
>
>
>
>
> "Martin 53N 1W" <ml_news@ddnospamddml1dd.co.uk.dd> wrote in message
> news:f6oMd.725$SC2.350@newsfe1-win.ntli.net...
> > Mike Schumann wrote:
> >> I am in the process of deploying VOIP at a number of office locations.
> >> My initial efforts have been somewhat disappointing, due to voice
quality
> >> problems.
> > [...]
> >
> > Look at what codecs you are using, what the link delay and jitter are,
and
> > work from that.
> >
> > You want minimum delay and jitter. Sometimes the lower bandwidth codecs
> > help in that you can run a link at lower than max capacity to improve
the
> > link performance.
> >
> > Don't share VoIP and internet traffic through the same bottleneck!!!
> >
> > ... And then you can play with VLANs and QoS...
> >
> > Good luck,
> > Martin
> >
> > --
> > ---------- OS? What's that?!
> > - Martin - To most people, "Operating System" is unknown & strange.
> > - 53N 1W - Mandrake 10.0.1 GNU Linux - An OS for Supercomputers & PCs
> > ---------- http://www.mandrakelinux.com/en-gb/concept.php3
>
>

Archived from groups: comp.dcom.voice-over-ip (More info?)

"Mike Schumann" <mike-nospam@traditions-nospam.com> wrote in message
news:b5mMd.7626$Ix.1914@newsread3.news.atl.earthlink.net...
> I am in the process of deploying VOIP at a number of office
> locations. My initial efforts have been somewhat disappointing,
> due to voice quality problems.

That may depend on insufficient bandwidth causing high dropout rate.
Mixing voice and data streams on the same link, without either plenty of
spare capacity or some QoS
(http://en.wikipedia.org/wiki/Quality_of_service ) mechanism to prioritize
the voice packets, is going to yield poor results.

> I am trying to get a better understanding of how SIP works, so
> I can figure out where the network bottlenecks are that are
> causing my voice problems.
>
> Does anyone know any good links to a detailed technical
> explanation of how > calls are handled by typical VOIP providers
> (Vonage, Broadvoice, etc...)?

Start here: http://www.iptel.org/ser/doc/sip_i [...] ction.html

> I am particularly interested in the question of how SIP proxies work
> (i.e. does all the voice traffic flow thru the proxies, or do they
> just initiate sessions and the two VOIP endpoints communicate directly).

SIP allows each endpoint to determine, among other parameters, the IP
address where to send the RTP stream. Usually this is the other endpoint's
address, but occasionally there could be other devices in the middle
(firewalls, NAT routers, RTP proxies etc.).

> Also, how do SIP
> terminal adapters work when they are located behind NAT routers?

It depends :-) . Read here:

http://www.newport-networks.com/wh [...] ersal.html
http://www.voip-info.org/wiki-ICE (ICE)
www.ietf.org/rfc/rfc3489.txt (STUN)
www.ietf.org/internet-drafts/draft [...] urn-06.txt (TURN)

Most modern devices implement STUN client capability, but as far as I know
very few support TURN/ICE.

Enzo

Archived from groups: comp.dcom.voice-over-ip (More info?)

Hank Karl wrote:
> On Thu, 03 Feb 2005 09:24:23 GMT, "Mike Schumann"
> <mike-nospam@traditions-nospam.com> wrote:
>
>
>>I am in the process of deploying VOIP at a number of office locations. My
>>initial efforts have been somewhat disappointing, due to voice quality
>>problems.
>
> Voice Quality impairments have a lot of different causes. Check out
> www.voiptroubleshooter.com

It covered quality of voice, but NOT Quality of Service (QoS) which I
think is the #1 problem when adding VoIP to existing systems. Ignoring
QoS is a major shortcoming of that web site.