If you experience consistent one way audio for each and every call, it is most likely to be a firewall issue. To troubleshoot, please check on the following:
- Double check the firewall settings and ensure that they follow the settings withing the SIP App configuration page. Ensure that your firewall settings is enabled for both incoming and outgoing. If you make any settings changes at this step, please remember to restart the firewall to ensure that the new settings are deployed.
Please note RTP may not come from the same IPs as the SIP signaling on our side. We do not recommend blocking RTP traffic in any way. We use UDP ports 10,000 to 30,000 for RTP. (See the Firewall setting in SIP App for more detail)
- If your SIP device is behind the NAT
- Please make sure NAT setting configured properly on SIP devices and Routers. The settings can be found within the SIP App configuration page. If you make any settings changes at this step, please remember to restart the IP-Phone and IP-PBX to ensure that the new settings are deployed.
- Confirm that SIP ALG is disabled. Many of today's lower end commercial routers implement SIP ALG coming with this feature enabled by default. While ALG could help in solving NAT related problems, the fact is that many routers' ALG implementations are wrong and break SIP. Hoiio will not support SIP ALG.
- Please consider setting for TURN & STUN server
- Confirm that you are using one of the supported codecs: G.711 (alaw, alaw), G.729, GSM. Consider testing with each other codecs (with ptime = 20ms)
Confirm your Network is in good condition (Sufficient Bandwidth, Latency, Packet Loss ...)