Hi,
We had modified some configuration in our cisco 800 series router. We set
all the UDP packets from our servers to ip precedence 5 and also allocate 75% of
bandwidth for UDP packets.
However we still facing latency and low volume problem. Is it our 512k
outbound bandwidth not enough to handle it? Thanks
Regards,
jorain
----- Original Message -----
From: jorain
To: asterisk-users at lists.digium.com
Sent: Sunday, December 09, 2007 5:56 PM
Subject: Re: [asterisk-users] asterisk performance
Thanks for your replies.
1.. Our connection mainly for voip, occasionally used for surfing websites.
2.. We are using codec g711u for local calls through TE120P, and g729 only if
making international calls through our sip provider, which only allow g723 and
g729. How can we get the license for g723? Which codec would you recommend?
3.. That quality problems we are facing are jitter, latency and occasionally
low volume. What cause these problems?
4.. No QoS Settings as we are quite new to it. Are we suppose to give high
priority to RTP in our router? What sort of QoS and traffic shapping would you
recommend?
5.. How many users can we expect to use voip(with good quality) with 512kbps
outbound connection?
Regards,
jorain
Date: Fri, 7 Dec 2007 10:27:36 -0500
From: "C F" <shmaltz at gmail.com>
Subject: Re: [asterisk-users] asterisk performance
To: jorain <jorain at caliber.com.sg>, "Asterisk Users Mailing List
-
Non-Commercial Discussion" <asterisk-users at lists.digium.com>
Message-ID:
<81000b5a0712070727s32156f31y4986abc9054144 at mail.gmail.com>
Content-Type: text/plain; charset=ISO-8859-1
by 3rd call do you mean over the internet?
if the answer is yes, then I wouldn't be surprised. another thing what
codec are you using?
Date: Fri, 7 Dec 2007 17:02:31 +0000
From: "Giovanni Miano" <giomiano at gmail.com>
Subject: Re: [asterisk-users] asterisk performance
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<asterisk-users at lists.digium.com>
Message-ID:
<d75be1ca0712070902u6d25ee49w368eda405a32bce8 at mail.gmail.com>
Content-Type: text/plain; charset=ISO-8859-1
2007/12/7, C F <shmaltz at gmail.com>:
> by 3rd call do you mean over the internet?
> if the answer is yes, then I wouldn't be surprised.
Oh my god!
If it is over internet and you get crap quality.. you have to be surprised..
It is depends by Latency (Traffic congestion, Network congestion) and
Packet loss
---------------------------------------------------------------------------------
jorain,
What do you mean for "quality problem" ?
Different "quality" problems are generated by different parameter
braking ? echo? low volume ?
Cheers
From: Michael Graves
To: Asterisk Users Mailing List - Non-Commercial Discussion ; jorain
Sent: Saturday, December 08, 2007 12:00 PM
Subject: Re: [asterisk-users] asterisk performance
Your 512k outbound bandwidth will tend to be the defining factor in call
quality here.
Does your connection only gets used for voip? Or is it shared with other uses?
Can you use more compressed codecs? G729 will quadruple you call capacity.
What sort of QoS and traffic shaping do you use? Note that these are separate
matters, and you need both.
Michael
--Original Message Text---
From: jorain
Date: Thu, 6 Dec 2007 17:47:18 +0800
Hi all,
We are using
- a dell sc440(Single dual-core intel xeon 3040, 1.86GHz,1066MHz front size
bus 2MB cache) as the asterisk server
- dell 400sc(Intel P4) as a SER server
- digium isdn card, TE120P at Asterisk server
- Bandwidth: 2Mbps/512kbps
All SIP Phones are registered to SER server, and SER will route all outgoing
calls to Asterisk server. My problem is the sound quality goes down if more than
3 concurent calls to PSTN.
Logically i think our system and bandwidth are more than enough to handle 3
concurent calls, but as the 4th person use it, the sound become jerky and a bit
delay. So how can we improve the sound quality?
Thanks
Regards,
jorain
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