Why one-way (or no-way) audio problem?
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.
Method 1: firewall
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.
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.
Method 2: STUN
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 'stun.counterpath.net' as STUN server.
If your MSS is deployed with public address, you can use it as your STUN server because it is embeded in MSS. Of course, you can also use our simple STUN server "stun.minisipserver.com".
By default, we suggest following STUN servers:
Method 3: relay media
In another way, we can try to configure MSS to relay media streams for SIP phones. In local users' configurations, please click tab "Media services" and enable "Relay media stream" item:
We need mention that all media streams will be sent MSS if you have configured such "relay media stream" item for local users. It will make heavy work-load to your server. And MSS cannot support relay video streams.
Method 4: avoid blocking
In some area, local ISPs could block or change SIP signals which can also cause one-way audio problem. To avoid that, you can try to change standard SIP port or use SIP over TCP. Please refer to following documents for more details.
Furthur more, we have two documents to describe more details of this topic. If you are interesting in it, please refer to following documents: