The following link provides excellent
indepth information regarding SIP Protocol and is presented in two
parts.Part one: http://arstechnica.com/business/news/2010/01/voip-in-depth-an-introduction-to-the-sip-protocol-part-1.ars Part
two: http://arste...
In version 7.0 or later
look at the "inspect" fields. (Prior to version 7 it was called "fixups"
not "inspect".) Make sure that the "inspect SIP" field is NO. "no
inspect sip" is the proper setting. ...
The iphone has known issues with DTMF: http://discussions.apple.com/message.jspa?messageID=6856277
The Linksys RV042 DOES
NOT SUPPORT the ability to change the UDP inactivity timeout. By default
the timeout is less than the 25 second keep-alive sends to keep the
port open.
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Polycom
IP Phone Setup Instructions
The
phone in the def...
Having the right router in your network to support good quality sound is
just half the job. You must set QoS to be High Priority on your router. These
steps will help you. First login to your WRP-400 router with the admin user and password. Look fo...
If you are a current AT&T customer and want to move your telephone service to Carolina Digital Phone you must first request your Customer Service Record (CSR). This information can be requested at the AT&T web site http://www.corp.att.com/lnp/...
Below is a list of
devices being used on our network that have been proven to work with
specific setting changes. Please be aware that there are many different
hardware and software versions, and not all versions are covered in this
document. The ...
Please be aware that we
do not support the use of SonicWall firewalls with the use of our
service. We have run through extensive testing with SonicWall's product
and it is not 100% efficient in handling SIP and can cause numerous call
routing and ...
System Upgrade Completed January 1, 2010, Problems discovered with Road Runner connections. A common problem we have discover today is with Time Warner Cable / Road Runner customers. This issue is that the Road Runner DNS is very slow to be updated. Some...
Problem: When I make a
call, the other party can't hear me, but I can hear them (or vice
versa). The typical situation is that you can be heard, but you cannot
hear the audio coming in the opposite direction.
Explanation: The probable cause of...
Quality of Service (QoS) is the secret sauce of many
custom VOIP solutions. The big bottleneck for VOIP QoS and the place
that causes most of the quality problems is the very first outbound
link. This is frequently the slowest link, being a DSL or ca...
Codec BR NEB G.729 8 Kbps 31.2 Kbps BR
=
Bit rate NEB = Nominal Ethernet Bandwidth (one direction)
GSM codec is indeed 13.2 kb/s (kilobits per
second). However when you want to
send the codec data across an IP network, it's going to increase the
used
bandwith. When dealing with voice you don't want to introduce too
much
latency. That's why...
SIP uses UDP port 5060 for signalling. The media is sent using RTP. Carolina Digital Phones' default configuration uses UDP ports 10,000 through 20,000 for RTP.
If you are getting dropped calls on a VoIP connection, it is most likely a problem with your network some where.
For one way audio on SIP phones make sure both
phones have 'canreinvite' set to no in sip.conf. Also make sure any NAT in front of the phones has ports
10,000 - 20,000 open.
In the analog phone world, FXS and FXO interfaces combine the receive and transmit signal on a single pair using a device called a hybrid. Because the transmitted and received signal are on the same pair, part of the transmitted signal comes back along wi...
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