<div dir="ltr">How can I set the maxptime value?<div><br></div><div>Setting the codec like this PCMA@8000h@40i?<br><div><br></div><div>Best regards</div></div></div><br><div class="gmail_quote"><div dir="ltr" class="gmail_attr">El lun, 19 sept 2022 a las 18:59, Sam via FreeSWITCH-users (<<a href="mailto:freeswitch-users@lists.freeswitch.org">freeswitch-users@lists.freeswitch.org</a>>) escribió:<br></div><blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex"><br><br><br>---------- Forwarded message ----------<br>From: Sam <<a href="mailto:sam4u@rediffmail.com" target="_blank">sam4u@rediffmail.com</a>><br>To: "<a href="mailto:freeswitch-users@lists.freeswitch.org" target="_blank">freeswitch-users@lists.freeswitch.org</a>" <<a href="mailto:freeswitch-users@lists.freeswitch.org" target="_blank">freeswitch-users@lists.freeswitch.org</a>><br>Cc: <br>Bcc: <br>Date: 19 Sep 2022 10:32:57 -0000<br>Subject: Re: [Freeswitch-users] RTP ptime changed multiple times<br>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 <<a href="mailto:josefu@gmail.com" target="_blank">josefu@gmail.com</a>><br>Sent: Mon, 19 Sep 2022 11:59:42 GMT+0200<br>To: FreeSWITCH Users Help <<a href="mailto:freeswitch-users@lists.freeswitch.org" target="_blank">freeswitch-users@lists.freeswitch.org</a>><br>Subject: Re: [Freeswitch-users] RTP ptime changed multiple times<br><br> <div style="display:block;background:rgb(97,97,97);border-radius:10px;padding:15px;float:left;margin:10px 15px 10px 0px"><p style="padding:0px;margin:0px;color:rgb(255,255,255);font-size:15px;line-height:30px"><b style="display:inline-block;margin:0px 10px 0px 0px;background:rgb(255,255,255);border-radius:100%;width:30px;height:30px;font-size:18px;font-family:serif;line-height:30px;text-align:center;color:rgb(97,97,97)">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 <<a href="mailto:josefu@gmail.com" target="_blank">josefu@gmail.com</a>> 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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<br><br><br><br>---------- Forwarded message ----------<br>From: Sam via FreeSWITCH-users <<a href="mailto:freeswitch-users@lists.freeswitch.org" target="_blank">freeswitch-users@lists.freeswitch.org</a>><br>To: "<a href="mailto:freeswitch-users@lists.freeswitch.org" target="_blank">freeswitch-users@lists.freeswitch.org</a>" <<a href="mailto:freeswitch-users@lists.freeswitch.org" target="_blank">freeswitch-users@lists.freeswitch.org</a>><br>Cc: <br>Bcc: <br>Date: Mon, 19 Sep 2022 09:59:29 -0700 (PDT)<br>Subject: Re: [Freeswitch-users] RTP ptime changed multiple times<br>_________________________________________________________________________<br>
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<a href="https://freeswitch.com" rel="noreferrer" target="_blank">https://freeswitch.com</a></blockquote></div><br clear="all"><div><br></div>-- <br><div dir="ltr" class="gmail_signature">Jose Fco. Irles Durá</div>