Hi All, We are using FreeSWITCH to put 2 members on a conference and record it. The problem is that when AMR-WB codec is used, the second member to join the conference hears distorted sound(sometimes it lasts couple seconds, in some cases it lasts until the end of the call). There is no such sound heard by first member participant, nor is it generated by it(verified by decoding audio from RTP packets sent from first member participant towards FreeSWITCH). Moreover, distorted sound to second member can be heard after decoding RTP packets sent from FreeSWITCH towards second conference participant - proof that it is not the issue with second participant phone or so. Intrestingly, there is no such issue when AMR codec is used(tested by removing AMR-WB from global_codec_prefs) - the sound is clear for both participant throughout the whole duration of the conference. Considering that everything is OK with AMR codec used, it seems like the issue is not with conference itself, only using AMR-WB during conference. We tested with some of the FreeSWITCH parameters related to RTP(e.g. auto-rtp-bugs, rtp-rewrite-timestamps, send_silence_when_idle, etc.) but it did not help. Is it known issue? Did anyone faced something similar before? What else can be done to make it work? Please, let me know if you have any questions regarding the topic/need more info from my side. Thanks, Rafal