<html><head><style>body{font-family:Helvetica,Arial;font-size:13px}</style></head><body style="word-wrap:break-word;line-break:after-white-space"><div id="bloop_customfont" style="font-family:Helvetica,Arial;font-size:13px;color:rgba(0,0,0,1.0);margin:0px;line-height:auto">Try <param name="rtp-rewrite-timestamps" value="true”/> in the sip profile.</div><div id="bloop_customfont" style="font-family:Helvetica,Arial;font-size:13px;color:rgba(0,0,0,1.0);margin:0px;line-height:auto"><br></div><div id="bloop_customfont" style="font-family:Helvetica,Arial;font-size:13px;color:rgba(0,0,0,1.0);margin:0px;line-height:auto">Regards,</div><div id="bloop_customfont" style="font-family:Helvetica,Arial;font-size:13px;color:rgba(0,0,0,1.0);margin:0px;line-height:auto"><br></div><div id="bloop_customfont" style="font-family:Helvetica,Arial;font-size:13px;color:rgba(0,0,0,1.0);margin:0px;line-height:auto">Naveen Khanna</div><div id="bloop_customfont" style="font-family:Helvetica,Arial;font-size:13px;color:rgba(0,0,0,1.0);margin:0px;line-height:auto"><br></div><p class="airmail_on">On 26 June 2018 at 6:37:06 AM, Sharath Kumar (<a href="mailto:shakumarsoftware@gmail.com">shakumarsoftware@gmail.com</a>) wrote:</p> <blockquote type="cite" class="clean_bq"><span><div><div></div><div>
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<div>Hello,<br>
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 ?<br></div>
Thank you,<br></div>
Shaks<br></div>
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