1. Home
  2. Voice-Over-IP
  3. SIP Technical Overview

SIP Technical Overview

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.

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

Thanks,

Reply to
Mike Schumann
Loading thread data ...

[...]

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

Reply to
Martin 53N 1W

i have found the following site is very useful:

formatting link

Reply to
DJC

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

formatting link
) mechanism to prioritize the voice packets, is going to yield poor results.

Start here:

formatting link

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.).

It depends :-) . Read here:

formatting link
(ICE)
formatting link
(STUN)
formatting link
(TURN)

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

Enzo

Reply to
Enzo Michelangeli

Why hasn't anyone mentioned

formatting link
It has EVERYTHING to do with VoIP, an excellent resource :)

-Paul

Reply to
Paul

You just did

Reply to
BlueRinse

Voice Quality impairments have a lot of different causes. Check out

formatting link
and

formatting link

Reply to
Hank Karl

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.

Reply to
Rick Merrill

Cabling-Design.com Forums website is not affiliated with any of the manufacturers or service providers discussed here. All logos and trade names are the property of their respective owners.