Audio Issues
When there are audio issues, the first response should be to determine where the client is sending the audio. To do this, check the SIP Dialog and SDP headers for where client was directed to. Wireshark is a valuable tool for such things.
For intra network calls between PCs, the audio SDP should most likely be pointing to a media proxy IP (unless STUN/TURN/ICE/etc is in use). If not, there could be some issue with media proxys (need more, none online, not configured correctly). Wireshark will also let you know where the client is really sending audio/RTP too.
For no audio:
- Check both User Agents are not muted and have speaker volume set correctly
- Check Calamar is configured with proper media proxy's
- Check Event log of media proxy to determine if Calamar/Sip Proxy is connected to it
- Check Performance Monitor (MMC) Snap-in for Remwave Media Proxy Counters to see if any audio/RTP data is going through. Possible to also use Wireshark
- Make sure Media Proxy is running and not firewalled (by Windows or other Firewalls)
- Ensure client is using a supported Codec
- Try both PC to PC and PC to PSTN calling to try to isolate problem location
Poor audio or noisy audio
- Ensure healthy network conditions (if your network is dropping lots of UDP traffic that is a problem)
- Ensure both ends are using correct CODECs
- If audio completely scrambled/gibberish at one side? - Check for mismatched codecs
The first place to start when nothing else has helped, is to locate the source of the problem. First, look at the SIP messages between User Agent and destination. Look at Codec's being used, look at audio ports & IPs and make sure correct audio stream is reaching. You can use wireshark to capture all audio/RTP streams (and even play back G711 Audio!). You can take the RTP streams and then use a tool like SIPp to broadcast the RTP to another user agent, there by reproducing the audio heard at each step of the way.
Major components responisble for Audio:
- User Agent
- Media Proxy for NAT'd calls
- TerraPBX Gateway - for PSTN calls
- Carrier Gateway
- General Network
Any one of those things can negatively affect the audio in a SIP communication if missconfigured or not functioning correctly.