We are often asked "why I cannot hear peer side?" or "why we cannot hear each other?". In most scenarios, the root reason is firewall filtering audio stream or NAT (network address translation) blocking it. We can always find that some SIP devices, including SIP phones, SIP clients and VoIP gateways, are deployed behind a NAT and configured with private IP address.
By the way, if your router can support ALG (Application Level Gateway) features, please check it and DISCARD all their SIP items since most routers have problems in processing SIP-ALG.
If there is a firewall deployed in your VoIP network, please try to shutdown it and make some test. If problem is resolved, that means you need ask administrator to open some ports for VoIP deployment. By default, VOIP will use following UDP ports: 5060, 5061, 10000~20000, and so on.
If your SIP phones/devices are deployed in a private network, mostly you need configure STUN(Simple Traversal of UDP through NATs) server to help your SIP devices to route packages, such as audio packages. Most SIP devices can support STUN protocol.
Below figure is STUN configuration in X-lite. By default, we configure its own 'stun.counterpath.net' as STUN server.
By default, we suggest following STUN servers:
With above configuration, if you still have one-way audio problem, you can try to configure "relay media stream" in local user configuration. Please refer to below figure. With this item, cloud-mss will relay its media streams (only support relaying audio stream). Since all audio streams will sent to or receive from cloud-mss who is in remote side, poor network connection could affect your audio quality.
We have two documents to describe more details about this topic. If you are interesting in it, please refer to following documents.