The following section outlines some common VoIP issues that may arise, and some recommended troubleshooting steps to narrow down the issue.
Audio only goes one way
Consider that voice communication typically happens as two simultaneous UDP streams, one for each direction of communication. These are two separate streams, as opposed to a single two-way stream. If communication in one direction is not making it to the peer, then the symptom is usually that only one party can hear the other's audio.
To address this issue, check the following:
- Trace the flow of traffic, and check any firewalls to ensure they are not blocking traffic.
- If a stateful firewall like the MX is passing traffic between the two peers, ensure there are appropriate mechanisms in place to allow inbound communication (1:1 NAT, port forwarding, etc).
- If it's unclear exactly where the traffic is being dropped, determine based on the symptoms which direction of traffic seems to be failing, and take packet captures at network hops to see where the flow stops.
Poor audio quality
Due to the sensitive nature of VoIP traffic, low voice quality (or "jitter") may be experienced due to interruptions in traffic flow or bandwidth limitations.
To improve voice quality, ensure that the following best practices are in place:
- Make sure voice traffic is segregated to its own voice VLAN, so normal data cannot interfere.
- Check the network's bandwidth limitations and ensure there's enough bandwidth (as recommended/required by the voice system).
- Use traffic shaping/QoS where necessary, in the event that a link on the network is being saturated.
- Take packet captures to get an idea of call quality, and where it is degraded. Many capture analysis tools, including Wireshark, have the ability to perform RTP analysis.
Take note of the "symptoms" exhibited in a poor-quality phone call. Specific traits of the call can help narrow down the issue. Please refer to this cisco guide for a breakdown of different call quality symptoms.
Phones can't get an IP address/configuration
Typically, VoIP equipment will get a dynamic configuration from a TFTP server or other service on the network. This will commonly be levied by a DHCP server, where leases to VoIP endpoints will include voice-specific DHCP options. In the event that the phone fails to connect to the network/get a working configuration, consider the following recommended steps:
- If a separate voice VLAN is being used, ensure that phones are being put on the appropriate VLAN by means of an access port, voice VLAN configuration on the port, or even configured on the phone itself.
- If this is being done, ensure that a DHCP server is up-and-running on that VLAN, and configured with the appropriate scope and options.
- If phones have a working IP configuration on the network (or a static assignment is given for test purposes) and are told to get their VoIP configuration from some other server, ensure that the server is online and reachable from the voice VLAN.