Thank you a lot for sharing the experience. I believe more users will take care to start separate instances for difference directionalstreaming when build a real VOIP solution based on the USC sample code.
Thank you for this important observation!ForP2P VoIPwith endpoint indifferent locationsa singlepair of encode and decodeinstancies canservetwo RTP streams: to encodeoutgoing and to decode incoming ones.Whilewithseveral incoming RTP streams (conference case)each endpointmust haveone independentdecodeinstance per stream.
Worth to note that the example code provided within USC manual (pdf)is not the onlyIPP speech-codec sample.
There isone more application like sample which thoughof course again is not a VoIP client or PBX one might expect from IPP,it is not usable as-is inreal network scenario like p2p communication or conference server.Nevertheless,umc_rtp_speech_codec sample is ready-to-build-and-executecode capable to encode to or decodefrom real RTP streams,for example captured using Wireshark andwhich we hope can be usefull at least in understanding how to build realthingusing USC codecs.