<html><head><meta http-equiv="content-type" content="text/html; charset=utf-8"></head><body dir="auto"><div dir="ltr">Hi Marek</div><div dir="ltr"><br></div><div dir="ltr">The way it works is that the sip message will carry the sdp and in that will be the instructions for where to route the rtp</div><div dir="ltr"><br></div><div dir="ltr">If systems don’t know their external address or are misconfigured they send an internal address that is not reachable from the far end. This is where lack of audio issues occur </div><div dir="ltr"><br></div><div dir="ltr">The main thing to show are the sip to/from addresses, anonymised but consistent, and the sdp data particularly the c and the media attributes</div><div dir="ltr"><br></div><div dir="ltr">Here is some background </div><div dir="ltr"><br></div><div dir="ltr"><a href="https://support.yeastar.com/hc/en-us/articles/360009806434-Brief-Introduction-of-SIP-and-SDP-Protocol-?mobile_site=true">https://support.yeastar.com/hc/en-us/articles/360009806434-Brief-Introduction-of-SIP-and-SDP-Protocol-?mobile_site=true</a></div><div dir="ltr"><br></div><div dir="ltr">If the c attribute has an address that is not routeable from the device that received it then you won’t get any audio</div><div dir="ltr"><br></div><div dir="ltr">It’s confusing at this stage but once you get a little more familiar it will start to make a lot of sense</div><div dir="ltr"><br><blockquote type="cite">On 5/09/2021, at 8:16 AM, Marek Greško <mgresko8@gmail.com> wrote:<br><br></blockquote></div><blockquote type="cite"><div dir="ltr"><span>Hello,</span><br><span></span><br><span>I agree my knowledge of SIP itself is poor, but I have quite well</span><br><span>general tcp/ip understanding. What sip parameters should be</span><br><span>anonymized? How about tag, branch, call-id, cseq values?</span><br><span></span><br><span>Thanks</span><br><span></span><br><span>Marek</span><br><span></span><br><span></span><br><span>2021-09-04 12:36 GMT+02:00, Duncan Turnbull <duncan@turnbull.co.nz>:</span><br><blockquote type="cite"><span></span><br></blockquote><blockquote type="cite"><span></span><br></blockquote><blockquote type="cite"><blockquote type="cite"><span>On 4/09/2021, at 8:55 PM, Marek Greško <mgresko8@gmail.com> wrote:</span><br></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"><span></span><br></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"><span>Ok,</span><br></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"><span></span><br></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"><span>let substitute lan for 192.168.100.235, provider with 192.0.2.1 and</span><br></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"><span>asterisk with 198.51.100.1.</span><br></blockquote></blockquote><blockquote type="cite"><span></span><br></blockquote><blockquote type="cite"><span>Can you provide the previous packet details with these addresses filled in</span><br></blockquote><blockquote type="cite"><blockquote type="cite"><span></span><br></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"><span>In the working scenario understand that asterisk is not aware of the</span><br></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"><span>providers ip address</span><br></blockquote></blockquote><blockquote type="cite"><span>If the call goes provider - asterisk - phone then asterisk is absolutely</span><br></blockquote><blockquote type="cite"><span>aware of the provider ip. I think you need to get more familiar with sip and</span><br></blockquote><blockquote type="cite"><span>rtp</span><br></blockquote><blockquote type="cite"><span></span><br></blockquote><blockquote type="cite"><blockquote type="cite"><span>192.0.2.1 in the sip protocol, and it should pick</span><br></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"><span>it from the network layer. It is harder to calcutale port, so it</span><br></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"><span>should probably listen for incoming rtp stream?</span><br></blockquote></blockquote><blockquote type="cite"><span></span><br></blockquote><blockquote type="cite"><span>The sdp in the sip packet tells the rtp ip and port to connect to</span><br></blockquote><blockquote type="cite"><blockquote type="cite"><span>Until then it is just</span><br></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"><span>sending to private address? But I thing it is futile, since it is</span><br></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"><span>known from the sip protocol there is nat involved and thus the packets</span><br></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"><span>are destined to nowhere.</span><br></blockquote></blockquote><blockquote type="cite"><span></span><br></blockquote><blockquote type="cite"><span>You need to realise that this works normally everyday all over the place so</span><br></blockquote><blockquote type="cite"><span>what you are imagining is incorrect</span><br></blockquote><blockquote type="cite"><blockquote type="cite"><span></span><br></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"><span>But I still cannot imagine what goes wrong in non working scenario and</span><br></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"><span>how the asterisk reboot (not every one and not sure this is the real</span><br></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"><span>trigger). The sip communication seems identical to me. The provider's</span><br></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"><span>router does not touch SIP now as observed after disabling SIP ALG.</span><br></blockquote></blockquote><blockquote type="cite"><span></span><br></blockquote><blockquote type="cite"><span>It is very unclear as to how you are justifying these statements. You don’t</span><br></blockquote><blockquote type="cite"><span>yet understand how sip and call setup with media works. If you provide the</span><br></blockquote><blockquote type="cite"><span>whole sip packet capture with the substituted ips it should be easier to</span><br></blockquote><blockquote type="cite"><span>point out where the error is</span><br></blockquote><blockquote type="cite"><span></span><br></blockquote><blockquote type="cite"><span>You need to be really clear on what’s ip</span><br></blockquote><blockquote type="cite"><span>is what and where the conversations are captured</span><br></blockquote><blockquote type="cite"><span></span><br></blockquote><blockquote type="cite"><span>It will become clear once you provide all the details</span><br></blockquote><blockquote type="cite"><span></span><br></blockquote><blockquote type="cite"><span></span><br></blockquote><blockquote type="cite"><blockquote type="cite"><span></span><br></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"><span>Thanks</span><br></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"><span></span><br></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"><span>Marek</span><br></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"><span></span><br></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"><span>2021-09-04 0:40 GMT+02:00, Antony Stone</span><br></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"><span><Antony.Stone@asterisk.open.source.it>:</span><br></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"><blockquote type="cite"><span>On Saturday 04 September 2021 at 00:34:49, Duncan Turnbull wrote:</span><br></blockquote></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"><blockquote type="cite"><span></span><br></blockquote></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"><blockquote type="cite"><blockquote type="cite"><blockquote type="cite"><blockquote type="cite"><span>On 4/09/2021, at 7:53 AM, Marek Greško <mgresko8@gmail.com> wrote:</span><br></blockquote></blockquote></blockquote></blockquote></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"><blockquote type="cite"><blockquote type="cite"><blockquote type="cite"><blockquote type="cite"><span></span><br></blockquote></blockquote></blockquote></blockquote></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"><blockquote type="cite"><blockquote type="cite"><blockquote type="cite"><blockquote type="cite"><span>So you suspect something is messing up SIP protocol? Maybe the phone</span><br></blockquote></blockquote></blockquote></blockquote></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"><blockquote type="cite"><blockquote type="cite"><blockquote type="cite"><blockquote type="cite"><span>itself is not working properly. The phone itself is not aware of the</span><br></blockquote></blockquote></blockquote></blockquote></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"><blockquote type="cite"><blockquote type="cite"><blockquote type="cite"><blockquote type="cite"><span>internet address, so is sending lan private address in the sip</span><br></blockquote></blockquote></blockquote></blockquote></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"><blockquote type="cite"><blockquote type="cite"><blockquote type="cite"><blockquote type="cite"><span>protocol.</span><br></blockquote></blockquote></blockquote></blockquote></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"><blockquote type="cite"><blockquote type="cite"><span></span><br></blockquote></blockquote></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"><blockquote type="cite"><blockquote type="cite"><span>I doubt it’s the phone. You need to check your data again and ideally</span><br></blockquote></blockquote></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"><blockquote type="cite"><blockquote type="cite"><span>explain what you mean by the names you have substituted for the ip</span><br></blockquote></blockquote></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"><blockquote type="cite"><blockquote type="cite"><span>addresses</span><br></blockquote></blockquote></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"><blockquote type="cite"><span></span><br></blockquote></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"><blockquote type="cite"><span>My advice (regarding the IP addresses) is:</span><br></blockquote></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"><blockquote type="cite"><span></span><br></blockquote></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"><blockquote type="cite"><span>- where you have https://tools.ietf.org/html/rfc1918 addresses, leave</span><br></blockquote></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"><blockquote type="cite"><span>them</span><br></blockquote></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"><blockquote type="cite"><span>as</span><br></blockquote></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"><blockquote type="cite"><span>they are - you're not giving away any sensitive information by telling us</span><br></blockquote></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"><blockquote type="cite"><span>about your internal addresses which can't be routed over the Internet</span><br></blockquote></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"><blockquote type="cite"><span></span><br></blockquote></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"><blockquote type="cite"><span>- where you have public addresses and would prefer not to reveal what</span><br></blockquote></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"><blockquote type="cite"><span>these</span><br></blockquote></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"><blockquote type="cite"><span>are, substitute with https://tools.ietf.org/html/rfc5737 addresses</span><br></blockquote></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"><blockquote type="cite"><span>instead.</span><br></blockquote></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"><blockquote type="cite"><span></span><br></blockquote></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"><blockquote type="cite"><span>- always ensure that you substitute address A in the same way each time,</span><br></blockquote></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"><blockquote type="cite"><span>and</span><br></blockquote></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"><blockquote type="cite"><span>address B, etc.</span><br></blockquote></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"><blockquote type="cite"><span></span><br></blockquote></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"><blockquote type="cite"><span></span><br></blockquote></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"><blockquote type="cite"><span>Antony.</span><br></blockquote></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"><blockquote type="cite"><span></span><br></blockquote></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"><blockquote type="cite"><span>--</span><br></blockquote></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"><blockquote type="cite"><span>You can spend the whole of your life trying to be popular,</span><br></blockquote></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"><blockquote type="cite"><span>but at the end of the day the size of the crowd at your funeral</span><br></blockquote></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"><blockquote type="cite"><span>will be largely dictated by the weather.</span><br></blockquote></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"><blockquote type="cite"><span></span><br></blockquote></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"><blockquote type="cite"><span>- Frank Skinner</span><br></blockquote></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"><blockquote type="cite"><span></span><br></blockquote></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"><blockquote type="cite"><span> Please reply to the</span><br></blockquote></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"><blockquote type="cite"><span>list;</span><br></blockquote></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"><blockquote type="cite"><span> please *don't* CC</span><br></blockquote></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"><blockquote type="cite"><span>me.</span><br></blockquote></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"><blockquote type="cite"><span></span><br></blockquote></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"><blockquote type="cite"><span>--</span><br></blockquote></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"><blockquote type="cite"><span>_____________________________________________________________________</span><br></blockquote></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"><blockquote type="cite"><span>-- Bandwidth and Colocation Provided by http://www.api-digital.com --</span><br></blockquote></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"><blockquote type="cite"><span></span><br></blockquote></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"><blockquote type="cite"><span>Check out the new Asterisk community forum at:</span><br></blockquote></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"><blockquote type="cite"><span>https://community.asterisk.org/</span><br></blockquote></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"><blockquote type="cite"><span></span><br></blockquote></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"><blockquote type="cite"><span>New to Asterisk? Start here:</span><br></blockquote></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"><blockquote type="cite"><span> https://wiki.asterisk.org/wiki/display/AST/Getting+Started</span><br></blockquote></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"><blockquote type="cite"><span></span><br></blockquote></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"><blockquote type="cite"><span>asterisk-users mailing list</span><br></blockquote></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"><blockquote type="cite"><span>To UNSUBSCRIBE or update options visit:</span><br></blockquote></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"><blockquote type="cite"><span> http://lists.digium.com/mailman/listinfo/asterisk-users</span><br></blockquote></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"><span></span><br></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"><span>--</span><br></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"><span>_____________________________________________________________________</span><br></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"><span>-- Bandwidth and Colocation Provided by http://www.api-digital.com --</span><br></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"><span></span><br></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"><span>Check out the new Asterisk community forum at:</span><br></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"><span>https://community.asterisk.org/</span><br></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"><span></span><br></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"><span>New to Asterisk? Start here:</span><br></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"><span> https://wiki.asterisk.org/wiki/display/AST/Getting+Started</span><br></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"><span></span><br></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"><span>asterisk-users mailing list</span><br></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"><span>To UNSUBSCRIBE or update options visit:</span><br></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"><span> http://lists.digium.com/mailman/listinfo/asterisk-users</span><br></blockquote></blockquote><blockquote type="cite"><span></span><br></blockquote><blockquote type="cite"><span>--</span><br></blockquote><blockquote type="cite"><span>_____________________________________________________________________</span><br></blockquote><blockquote type="cite"><span>-- Bandwidth and Colocation Provided by http://www.api-digital.com --</span><br></blockquote><blockquote type="cite"><span></span><br></blockquote><blockquote type="cite"><span>Check out the new Asterisk community forum at:</span><br></blockquote><blockquote type="cite"><span>https://community.asterisk.org/</span><br></blockquote><blockquote type="cite"><span></span><br></blockquote><blockquote type="cite"><span>New to Asterisk? Start here:</span><br></blockquote><blockquote type="cite"><span> https://wiki.asterisk.org/wiki/display/AST/Getting+Started</span><br></blockquote><blockquote type="cite"><span></span><br></blockquote><blockquote type="cite"><span>asterisk-users mailing list</span><br></blockquote><blockquote type="cite"><span>To UNSUBSCRIBE or update options visit:</span><br></blockquote><blockquote type="cite"><span> http://lists.digium.com/mailman/listinfo/asterisk-users</span><br></blockquote><span></span><br><span>-- </span><br><span>_____________________________________________________________________</span><br><span>-- Bandwidth and Colocation Provided by http://www.api-digital.com --</span><br><span></span><br><span>Check out the new Asterisk community forum at: https://community.asterisk.org/</span><br><span></span><br><span>New to Asterisk? Start here:</span><br><span> https://wiki.asterisk.org/wiki/display/AST/Getting+Started</span><br><span></span><br><span>asterisk-users mailing list</span><br><span>To UNSUBSCRIBE or update options visit:</span><br><span> http://lists.digium.com/mailman/listinfo/asterisk-users</span></div></blockquote></body></html>