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 -- Michael Graves mgraves<at>mstvp.com o713-861-4005 c713-201-1262 sip:[EMAIL PROTECTED] skype mjgraves fwd 54245
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