Question about VoIP

svt11

Occasional Visitor
Hi, one of my clients is call center which make calls in Turkey. They use Bria and Zoiper as softphones and their VoIP provider is located in UK or Israel. They use UK numbers and the call center is actually in Bulgaria. So, I'm new to VoIP and I know it's latency sensitive. Even with CCNA certificate I still don't understand good latency. But like with online games, calls are latency-sensitive. So when using UK number incoming calls must be routed from Turkey through UK and then to Bulgaria, right? I understand that I need symmetrical internet. There are max 35 phones used in callcenter and ofcourse there are other computers which are using internet for other purposes. There is ISP which doesn't give such connection. Latency to UK is 40-50ms, upload speeds are about 25-40mbit. Problems which we have are calls being interrupted after 30sec, but it was from Bria. I think I've fixed it. Also another problem - other side not hearing us, or one-way call and another is repeating of the call from the biginning. Colleagues say they hear once again what the client said from the beginiing - they hear first 30 secs for example once again which tells me that packets are being retransmitted from the other side again. Soon, I will buy internet from ISP which is symmtrical and is TIER 2 ISP. So latency will be slower. I may do another thing - prioritize VoIP in my central router which Mikrotik. Maybe this will handle the things more better. I'm looking now to replace the VoIP provider and his location to be in Bulgaria.
 

Cake

Senior Member
Latency of 40-50ms is good IMO. Sound like your clients are not configured correctly. When you say ..other computers which are using Internet for other purposes ... I think that could be a issue if you also have 35 voip phones. With the 'other side not hearing us' is usually a configuration issue I have also had (client).

I know your post is a little old, but if your still looking for a fix try using a better codec. If I understand you, your making voip to a PSTN? I have good experience with didlogic, also you could run a server- kamailio + rtpproxy on your network and link that to a PSTN, but IMO it has a good learning curve. 35 phones seems like a lot to configure.
 

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