Thanks Naveen! I will try this but I thought that was only for packets that FS generates. On Mon, Jun 25, 2018 at 11:22 PM, Naveen Khanna <naveen.khanna.bm@gmail.com> wrote:
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
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