In this article I am going to cover some simple and common steps that a user of a VoIP/SIP telephony system can do before calling they VoIP provider.
No inbound calls:
- Firewall – Make sure the firewall is not blocking inbound connection to the phone/PBX. As a general rule make sure that UDP ports 5060, 10000-20000 are open for SIP/RTP Streams. You might also need to check other Audio RTP Ports based on the phone/Equitment you are using.
- Broadband – Make sure that your Broadband provider is not blocking ports or the SIP service to and from your VoIP provider.
One way audio:
- Firewall Ports – Make sure the firewall is not blocking ports, as mentioned above.
- Codec Settings – Make sure you are using the correct Codec settings. A lot of providers prefer you to use G.711Alaw. using the wrong codec on any port of the call can stop audio from working in a direction if the VoIP provider is not Transcoding between the codecs.
Bad Audio/Darlek Sounding:
- Check LAN – Make sure you have no internal LAN problems like Loops or programs that send out high volumes of multi-cast packets.
- Check Bandwidth – Make sure that on both your LAN and WAN/ISP you have enough bandwidth to suit the codec’s you are using for SIP/VoIP. As a rule, G.711Alaw uses 87.2kbps(8.7KBps). Also make sure you have no users on the network using Bit torrent clients for seeding and downloading
- Check Broadband – Check the Bandwidth you have available from your ISP to make sure you can make calls with a guaranteed bandwidth amount. As a rule you want a minimum of 2Mb down, and 2Mb up.
- Check For Packet Loss – run a packet loss trace using either MTR (Linux) or WinMTR (Windows) to both your ISPs DNS Servers AND the VoIP providers servers you are connecting to, if you see any lost packets, then you know that your calls are going to have words and bits missing, and you will be either asking to repeat stuff or be asked to repeat stuff.