<html><body><div style="font-family: arial, helvetica, sans-serif; font-size: 12pt; color: #000000"><div>You could only reduce the packet size using headers compresions, reuducing SDP offers or a combination of both but I'm sure they will not support that. VoIPStunt it's not a 'serious' carrier from you could spect support from their side on this.<br></div><div><br data-mce-bogus="1"></div><div>Also it's strange that you packet get oversized if you have only 2 RR<br data-mce-bogus="1"></div><div><br data-mce-bogus="1"></div><div>About the no RTP-flow on internal calls ... check using sngrep the full-flow and see whats getting wrong when negotiated.<br data-mce-bogus="1"></div><div><br data-mce-bogus="1"></div><div><br></div><hr id="zwchr" data-marker="__DIVIDER__"><div data-marker="__HEADERS__"><blockquote style="border-left: 2px solid #1010FF; margin-left: 5px; padding-left: 5px; color: #000; font-weight: normal; font-style: normal; text-decoration: none; font-family: Helvetica,Arial,sans-serif; font-size: 12pt;" data-mce-style="border-left: 2px solid #1010FF; margin-left: 5px; padding-left: 5px; color: #000; font-weight: normal; font-style: normal; text-decoration: none; font-family: Helvetica,Arial,sans-serif; font-size: 12pt;"><b>De: </b>"Jonathan Yue" <jonathan.yue@turboitsolutions.com><br><b>Para: </b>"Raúl Alexis Betancor Santana" <rabs@dimension-virtual.com>, spce-user@lists.sipwise.com<br><b>Enviados: </b>Jueves, 14 de Abril 2016 4:26:53<br><b>Asunto: </b>Re[2]: [Spce-user] sip fragmentation<br></blockquote></div><div data-marker="__QUOTED_TEXT__"><blockquote style="border-left: 2px solid #1010FF; margin-left: 5px; padding-left: 5px; color: #000; font-weight: normal; font-style: normal; text-decoration: none; font-family: Helvetica,Arial,sans-serif; font-size: 12pt;" data-mce-style="border-left: 2px solid #1010FF; margin-left: 5px; padding-left: 5px; color: #000; font-weight: normal; font-style: normal; text-decoration: none; font-family: Helvetica,Arial,sans-serif; font-size: 12pt;"><div>I applied that workaround: <a href="https://www.sipwise.org/doc/2.8/spce/ar01s05.html#_audiocodes_devices_workaround" target="_blank" data-mce-href="https://www.sipwise.org/doc/2.8/spce/ar01s05.html#_audiocodes_devices_workaround">https://www.sipwise.org/doc/2.8/spce/ar01s05.html#_audiocodes_devices_workaround</a>, from the packet captures, I can see 127.0.0.1 in RR header is changed to a private ip address. but issue persists.</div><div> </div><div>So I went back to the thought of reducing sdp <span id="xdaf6d33443ec489dac07b183dd2dcdee">size</span>. on softphone MicroSIP, I disabled ICE, enabled only 1 codec. bingo! the sdp is now small enough to be contained in 1 packet. then the outbound call via VoIPStunt succeeded! this test proved my thought about packet fragmentation, but I still wonder what is the official solution for such situation? to disable ICE and keep only 1 codec isn't nice way to configure a sip client.</div><div> </div><div>Now the new problem is that there's no rtp traffics between in internal call.</div><div> </div><div>------ Original Message ------</div><div>From: "Raúl Alexis Betancor Santana" <<a href="mailto:rabs@dimension-virtual.com" target="_blank" data-mce-href="mailto:rabs@dimension-virtual.com">rabs@dimension-virtual.com</a>></div><div>To: <a href="mailto:spce-user@lists.sipwise.com" target="_blank" data-mce-href="mailto:spce-user@lists.sipwise.com">spce-user@lists.sipwise.com</a><br data-mce-bogus="1"></div><div>Sent: 2016-04-13 4:11:29 PM</div><div>Subject: Re: [Spce-user] sip fragmentation</div><div> </div><div id="x33709d2746514b589eb0686777e3f638"><blockquote class="cite2" cite="1718805197.76221.1460589089715.JavaMail.zimbra@dimension-virtual.com"><div style="font-size: 12pt; font-family: arial, helvetica, sans-serif; color: #000000;" data-mce-style="font-size: 12pt; font-family: arial, helvetica, sans-serif; color: #000000;"><div>Trust me ... it's not a fragmentation problem ... just follow the indications on the documentation about the "Audiocodes devices workaround", the real problem it's the 1.27.0.0.1 IP on the RR headers ... VoIPStunt and a lot of other SIP-Carriers, doesn't support private IP's on the SIP headers.<br></div><br> <br><div><hr id="zwchr"></div><div><blockquote style="font-size: 12pt; text-decoration: none; font-family: Helvetica,Arial,sans-serif; font-weight: normal; color: #000; font-style: normal; padding-left: 5px; margin-left: 5px; border-left: #1010ff 2px solid;" data-mce-style="font-size: 12pt; text-decoration: none; font-family: Helvetica,Arial,sans-serif; font-weight: normal; color: #000; font-style: normal; padding-left: 5px; margin-left: 5px; border-left: #1010ff 2px solid;"><b>De: </b>"Jonathan Yue" <<a href="mailto:jonathan.yue@turboitsolutions.com" target="_blank" data-mce-href="mailto:jonathan.yue@turboitsolutions.com">jonathan.yue@turboitsolutions.com</a>><br><b>Para: </b>"Raúl Alexis Betancor Santana" <<a href="mailto:rabs@dimension-virtual.com" target="_blank" data-mce-href="mailto:rabs@dimension-virtual.com">rabs@dimension-virtual.com</a>>, <a href="mailto:spce-user@lists.sipwise.com" target="_blank" data-mce-href="mailto:spce-user@lists.sipwise.com">spce-user@lists.sipwise.com</a><br><b>Enviados: </b>Miércoles, 13 de Abril 2016 23:09:45<br><b>Asunto: </b>Re[2]: [Spce-user] sip fragmentation<br></blockquote></div><div><blockquote style="font-size: 12pt; text-decoration: none; font-family: Helvetica,Arial,sans-serif; font-weight: normal; color: #000; font-style: normal; padding-left: 5px; margin-left: 5px; border-left: #1010ff 2px solid;" data-mce-style="font-size: 12pt; text-decoration: none; font-family: Helvetica,Arial,sans-serif; font-weight: normal; color: #000; font-style: normal; padding-left: 5px; margin-left: 5px; border-left: #1010ff 2px solid;"><div>Thanks for your input Raúl.</div><div> </div><div>in the packet captures, I notice following items:</div><div>INVITE sip:<a href="mailto:1xxxxxxxxxx@sip.voipstunt.com" target="_blank" data-mce-href="mailto:1xxxxxxxxxx@sip.voipstunt.com">1xxxxxxxxxx@sip.voipstunt.com</a> SIP/2.0<br>Record-Route: <sip:x.x.x.115;r2=on;lr=on;ftag=3F30077D-570BCE0A000B61B3-54909700;ngcplb=yes><br>Record-Route: <sip:127.0.0.1;r2=on;lr=on;ftag=3F30077D-570BCE0A000B61B3-54909700;ngcplb=yes><br>Via: SIP/2.0/UDP x.x.x.115;branch=z9hG4bK037e.d0ff913568a8906a194cc8f2492a77c7.0<br>Via: SIP/2.0/UDP 127.0.0.1:5080;received=127.0.0.1;branch=z9hG4bKsdiM4aEk;rport=5080</div><div> </div><div>I wonder if the 2nd record-route message and 2nd Via message are necessary. If they are omitted, the sip packet won't be fragmented.</div><div> </div><div>------ Original Message ------</div><div>From: "Raúl Alexis Betancor Santana" <<a href="mailto:rabs@dimension-virtual.com" target="_blank" data-mce-href="mailto:rabs@dimension-virtual.com">rabs@dimension-virtual.com</a>></div><div>To: <a href="mailto:spce-user@lists.sipwise.com" target="_blank" data-mce-href="mailto:spce-user@lists.sipwise.com">spce-user@lists.sipwise.com</a><br></div><div>Sent: 2016-04-13 1:02:40 AM</div><div>Subject: Re: [Spce-user] sip fragmentation</div><div> </div><div id="xcf4f1dc3d143449a82786539bffdd321"><blockquote class="cite2" cite="615346034.70358.1460534560937.JavaMail.zimbra@dimension-virtual.com"><div style="font-size: 12pt; font-family: arial, helvetica, sans-serif; color: #000000;" data-mce-style="font-size: 12pt; font-family: arial, helvetica, sans-serif; color: #000000;"><div>You could activate header compresion, but you will have to pray to VoIPStunt to support that.<br></div><br><div>Better check why your headers get so big<br></div><br><div><div><hr id="zwchr"></div></div><div><blockquote style="font-size: 12pt; text-decoration: none; font-family: Helvetica,Arial,sans-serif; font-weight: normal; color: #000; font-style: normal; padding-left: 5px; margin-left: 5px; border-left: #1010ff 2px solid;" data-mce-style="font-size: 12pt; text-decoration: none; font-family: Helvetica,Arial,sans-serif; font-weight: normal; color: #000; font-style: normal; padding-left: 5px; margin-left: 5px; border-left: #1010ff 2px solid;"><b>De: </b>"Jonathan Yue" <<a href="mailto:jonathan.yue@turboitsolutions.com" target="_blank" data-mce-href="mailto:jonathan.yue@turboitsolutions.com">jonathan.yue@turboitsolutions.com</a>><br><b>Para: </b><a href="mailto:spce-user@lists.sipwise.com" target="_blank" data-mce-href="mailto:spce-user@lists.sipwise.com">spce-user@lists.sipwise.com</a><br><b>Enviados: </b>Martes, 12 de Abril 2016 23:48:43<br><b>Asunto: </b>[Spce-user] sip fragmentation<br></blockquote></div><div><blockquote style="font-size: 12pt; text-decoration: none; font-family: Helvetica,Arial,sans-serif; font-weight: normal; color: #000; font-style: normal; padding-left: 5px; margin-left: 5px; border-left: #1010ff 2px solid;" data-mce-style="font-size: 12pt; text-decoration: none; font-family: Helvetica,Arial,sans-serif; font-weight: normal; color: #000; font-style: normal; padding-left: 5px; margin-left: 5px; border-left: #1010ff 2px solid;"><div>Hi, gang,</div><div> </div><div>Is there a way to shrink the header of sip message sent from spce? I configured a <span id="x761f1c3b6ea84d86ada77d3cd52807cc">sip trunk with </span>VoIPStunt as a test. After receiving invite message, VoIPStunt sends 401, spce then sends out an sdp, supposedly containing authentication digest, however, the message gets fragmented. VoIPStung never responds again, apparently it doesn't assemble fragmented packets.</div><div> </div><div>I then change the trunk's transport protocol to tcp, VoIPStunt doesn't respond at all, I guess it doesn't support tcp.</div><div> </div><div>spce version is v4.2.1. in the attached screenshot, 77.72.169.129 is VoIPStunt's ip address, the other is my spce's.</div><div> </div><div>thanks,</div><div> </div><div>Jon</div><br>_______________________________________________<br>Spce-user mailing list<br><a href="mailto:Spce-user@lists.sipwise.com" target="_blank" data-mce-href="mailto:Spce-user@lists.sipwise.com">Spce-user@lists.sipwise.com</a><br><a href="https://lists.sipwise.com/listinfo/spce-user" target="_blank" data-mce-href="https://lists.sipwise.com/listinfo/spce-user">https://lists.sipwise.com/listinfo/spce-user</a><br></blockquote></div></div></blockquote></div></blockquote></div></div></blockquote></div><br></blockquote></div></div></body></html>