FAQ: Message Control || VoIP message is sent but can’t be heard

FAQ: Message Control || VoIP message is sent but can’t be heard

In a VoIP message the audio file can use several codec formats for its compression.

When a call is set up, then sometimes some target providers already play back the ringing to the call as an audio stream. This can be done in several or one specific codec. A VoIP service forward the codecs from the destination to the caller (Service Engine)  and vice versa on a one-to-one basis, it can be the case that the system receives one or more codecs.
It is also possible to switch codecs, so that the caller receives the ringing in an audio stream via PCMU for example, but the actual call is then answered with PCMA and the stream also switches from PCMU to PCMA. 

In zenon the higher ranked codec is the PCMU, also, zenon doesn’t have the capability to change the codec used after the RTP session is initiated. This means that if the codec is changed in the destination for playback, for example, to PCMA or G722, the codec associated to the message sent by zenon cannot be changed anymore and the playback will not work as expected.

It is possible to find which codec is being used via a wireshark capture, in the the packets from the SIP/SDP Protocol, eg:


List of some commonly used codecs:
  1. G.711 (PCMA, PCMU): Traditional, high-quality codec, widely compatible with telephony systems. Most European VoIP systems use this;
  2. G.729: Efficient codec for bandwidth-constrained environments, popular in enterprise VoIP;
  3. G.722: Wideband codec for HD voice, gaining popularity in VoIP services;
  4. AMR-WB: Wideband codec commonly used in mobile networks;
  5. iLBC: Robust codec for lossy network conditions, used in some VoIP applications.


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