Acquiring means for acquiring files containing degraded audio signals from a Secure Real-Time Transport Protocol, SRTP, -stream, the files being acquired for an objective audio quality assessment method, wherein traffic with the SRTP stream is transmitted during a successfully established Web Real-Time Communication, WebRTC, audio media call between at least two WebRTC endpoints (1a, 1b), wherein the at least two WebRTC endpoints (1a, 1b) are network elements of a network, wherein the SRTP stream is transmitted via a WebRTC audio media transmission path between a sending endpoint of the at least two WebRTC endpoints (1a, 1b) and a receiving endpoint of the at least two WebRTC endpoints (1a, 1b); at least one further network element different from the WebRTC endpoints (1a, 1b), wherein the at least one further network element is located at the WebRTC audio media transmission path, wherein the traffic is transmitted by the at least one further network element via the WebRTC audio media transmission path; the acquiring means comprising: receiving means for receiving information about the WebRTC endpoints (1a, 1b) and reference information, wherein the information about the WebRTC endpoints (1a, 1b) and the reference information comprise an encryption key, a cipher suite and a Synchronization Source, SSRC, identifier; reading means for reading at least one audio stream from at least one previously captured SRTP traffic file, wherein the at least one previously captured SRTP traffic file containing the at least one audio stream is captured from at least one previous WebRTC audio media call; parsing means for parsing the information about the WebRTC endpoints (1a, 1b) and the reference information; filtering means for receiving the traffic and for filtering the SRTP stream from the traffic, wherein the filtering means is configured to filter the SRTP stream according to the SSRC identifier obtained from the information about the WebRTC endpoints (1a, 1b) and the reference information and according to a simulation of behavior of a jitter buffer of the WebRTC receiving endpoint of the at least two WebRTC endpoints (1a, 1b); decryption means for performing SRTP decryption with the encryption key and the cipher suite in order to generate decrypted Real-Time Transport Protocol, RTP, packets; extracting means for extracting RTP payload from the decrypted RTP packets, wherein the RTP payload contains an encoded, degraded audio signal; decoding means for decoding the encoded, degraded audio signal, wherein the decoding means comprises an audio decoder configured to produce decoded Pulse Code Modulation, PCM, -audio streams; preparing means for preparing the decoded PCM audio streams according to requirements of an objective audio quality assessment method, wherein the preparing means is configured to split the decoded PCM audio streams into multiple decoded PCM audio samples; output means for outputting at least one decoded PCM audio sample of the multiple decoded PCM audio samples in a file format, wherein the file format is appropriate to the requirements of the objective audio quality assessment method. receiving means for receiving information about the WebRTC endpoints (1a, 1b) and reference information, wherein the information about the WebRTC endpoints (1a, 1b) and the reference information comprise an encryption key, a cipher suite and a Synchronization Source, SSRC, identifier; reading means for reading at least one audio stream from at least one previously captured SRTP traffic file, wherein the at least one previously captured SRTP traffic file containing the at least one audio stream is captured from at least one previous WebRTC audio media call; parsing means for parsing the information about the WebRTC endpoints (1a, 1b) and the reference information; filtering means for receiving the traffic and for filtering the SRTP stream from the traffic, wherein the filtering means is configured to filter the SRTP stream according to the SSRC identifier obtained from the information about the WebRTC endpoints (1a, 1b) and the reference information and according to a simulation of behavior of a jitter buffer of the WebRTC receiving endpoint of the at least two WebRTC endpoints (1a, 1b); decryption means for performing SRTP decryption with the encryption key and the cipher suite in order to generate decrypted Real-Time Transport Protocol, RTP, packets; extracting means for extracting RTP payload from the decrypted RTP packets, wherein the RTP payload contains an encoded, degraded audio signal; decoding means for decoding the encoded, degraded audio signal, wherein the decoding means comprises an audio decoder configured to produce decoded Pulse Code Modulation, PCM, -audio streams; preparing means for preparing the decoded PCM audio streams according to requirements of an objective audio quality assessment method, wherein the preparing means is configured to split the decoded PCM audio streams into multiple decoded PCM audio samples; output means for outputting at least one decoded PCM audio sample of the multiple decoded PCM audio samples in a file format, wherein the file format is appropriate to the requirements of the objective audio quality assessment method.