Hello Jose<br /><br />Sorry to interrupt the thread, but just wondering if you think adding a maxptime attribute in the SDP may be a work around for you?<br />If you set this to 20, the switch between 20/40 should not happen.<br /><br />Kind regards<br /><br />Sameee<br /><br/><br><br><br><br>From: Jose Fco. Irles Dur <josefu@gmail.com><br>Sent: Mon, 19 Sep 2022 11:59:42 GMT+0200<br>To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org><br>Subject: Re: [Freeswitch-users] RTP ptime changed multiple times<br><br> <div style="display:block; background:#616161; border-radius:10px; padding:15px; width:calc(100% - 45px); float:left; margin:10px 15px 10px 0;"><p style="padding:0;margin:0; color:#fff; font-size:15px; line-height:30px; "><b style="display:inline-block; margin:0 10px 0 0; background:#fff; border-radius:100%; width:30px; height:30px; font-size:18px; font-family:serif; line-height:30px; text-align:center; color:#616161;">i</b>We couldn't verify the sender of this email.</p></div><div dir="ltr"><div class="gmail_quote"><blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex"><div dir="ltr"><br>Tell the other side to fix their brokenness maybe? How did you try setting the flag?<div><br></div></div></blockquote><div><br></div><div>Yes, I know that the problem is from the provider, but its response time is bad.</div><div><br></div><div>I setted "rtp_media_autofix_timing=false" for testing, but when I setted, the first packets (40ms) are ignored and when the ptime is 20ms I listen to it. I think this is the correct behavior according to my understanding, so I can't solve the problem with the buggy provider.</div><div><br></div><div>Best regards</div><div><br></div><div> </div><blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex"><div dir="ltr"><div></div><div>/b</div><div><br></div></div><br><div class="gmail_quote"><div dir="ltr" class="gmail_attr">On Fri, Sep 16, 2022 at 4:28 PM Jose Fco. Irles Durá <josefu@gmail.com> wrote:<br></div><blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex"><div dir="ltr">Hello, I have a problem with some calls where the rtp stream from the other side changes the rtp ptime multiple times.<div><br></div><div>I make a call with the sdp line "a=ptime:20" but the initial rtp comes with 40ms ptime.</div><div>FreeSWITCH detects it and changes the ptime to 40ms and all works ok.</div><div>Later in the conversation, without any re-invite, the rtp stream changed to 20ms. At this moment, FreeSWITCH does not detect the change and the audio sended by FreeSWITCH to the other legs is choppy.</div><div><br></div><div>I think that the problem is on the other side, but the other side is a big operator and always this is a problem to solve issues.</div><div><br></div><div>I tried to set "rtp_media_autofix_timing", but if I set this property, the first part of the conversation is discarded (ptime at 40ms, silence in my phone) and later I can listen the audio (when the ptime=20ms).</div><div><br></div><div>Is there any solution for this?</div><div><br></div><div>Best regards<br clear="all"><div><br></div>-- <br><div dir="ltr">Jose Fco. Irles Durá</div></div></div>
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