Try <param name="rtp-rewrite-timestamps" value="true”/> in the sip profile. Regards, Naveen Khanna On 26 June 2018 at 6:37:06 AM, Sharath Kumar (shakumarsoftware@gmail.com) wrote: Hello, I am running FS 1.6.18 and have a provider that is sending me a stream of incoming RTP that changes it's host IP, and RTP sequence number and Timestamps after 10 mins in the call. The SSRC does not change. As a result I am getting either silence or garbled audio. The Sample rate is 20ms as in the original stream. Is there any setting in the FS that can make this work ? Have any of seen this work with inbound stream ? Thank you, Shaks _________________________________________________________________________ Professional FreeSWITCH Services sales@freeswitch.com https://freeswitch.com Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users https://freeswitch.com