We are often asked "why I cannot hear peer side?" or "why we cannot hear each other?". In most scenarios, the root reason is 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 router and configured with private IP address.
Please check your router firstly. If it can support ALG (Application Level Gateway) features, please DISCARD all their SIP items since most routers have problems in processing SIP-ALG.
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.
For example, microSIP has this feature and you can refer to following figure to configure its settings.
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.