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

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