<div dir="ltr">Hi Andrew and All<div><br></div><div>I changed <span style="font-size:12.8px">strict_routing_safe: 'yes' and applied but it does't solve my problem, the lb still drops ACK package from Cisco gateway</span></div><div><br></div><div>Here is the log from kamailio-lb.log </div><div><br></div><div><div><div><i>Sep 25 19:25:54 spce lb[8628]: NOTICE: <script>: New request on lb - M=ACK R=sip:x.x.x.x:6060;r2=on;lr=on;ftag=74E76334-5D1;nat=yes;ngcplb=yes;socket=udp:x.x.x.x:6060 F=sip:2818573448@y.y.y.y T=sip:17133751530@x.x.x.x IP=udp:y.y.y.y:56266 ID=C476CBE2-307B11D5-9770AB03-128F69C7@y.y.y.y UA='<null>'</i></div><div><i>Sep 25 19:25:54 spce lb[8628]: WARNING: <script>: Last hop of ACK points to 'x.x.x.x:6060' which is not a proxy, drop - R=sip:127.0.0.1:5080;prxroute=1 ID=C476CBE2-307B11D5-9770AB03-128F69C7@y.y.y.y UA='<null>'</i></div></div></div><div><br></div><div>There is not much thing I can do on Cisco's configuration to change its behaviors so there is anyway we can make the kaimailio-lb to accept that ACK and forward to proxy?</div><div><br></div><div>Thank you all </div><div><span style="font-size:12.8px"><br></span></div><div><span style="font-size:12.8px"><br></span></div></div><div class="gmail_extra"><br clear="all"><div><div class="gmail_signature"><div dir="ltr"><br><div>---</div><div>Best regards,</div><div><br></div><div><b>Tung Tran</b></div><div><br></div><div><div><font color="#0000ff"><b>V247 Enterprise Corp<br></b>713.358.2257 office | 281.857.3448 cell<br>9999 Bellaire Blvd., Ste. 1111<b> | </b>Houston, TX 77036<br></font></div><div><b><font color="#0000ff">tung.tran@V247.com | <a href="http://www.v247.com/" target="_blank">www.V247.com</a></font></b></div></div><div><br></div></div></div></div>
<br><div class="gmail_quote">On Fri, Sep 25, 2015 at 5:33 AM, Andrew Pogrebennyk <span dir="ltr"><<a href="mailto:apogrebennyk@sipwise.com" target="_blank">apogrebennyk@sipwise.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">Hi Tung,<br>
you may try to enable the parameter<br>
strict_routing_safe: 'yes'<br>
in /etc/ngcp-config/config.yml and do ngcpcfg apply..<br>
The system will do its best to try to route the stict-routed packets.<br>
<br>
Regards,<br>
Andrew<br>
<div><div class="h5"><br>
On 09/25/2015 08:09 AM, Daniel Grotti wrote:<br>
> Looks like he's doing strict routing an not loose routing. He's setting<br>
> the route value as ruri, and not the contact in 200ok.<br>
> It is not following rfc3261.<br>
><br>
> Daniel<br>
><br>
> On Sep 24, 2015 11:55 PM, Tung Tran <<a href="mailto:tung.tran@v247.com">tung.tran@v247.com</a>> wrote:<br>
><br>
> Hi Daniel<br>
><br>
> Here is 200 OK and ACK from Cisco gateway<br>
><br>
> x.x.x.x: SPCE<br>
> y.y.y.y: Cisco GW<br>
><br>
><br>
> Apr 14 05:54:00.713: Received:<br>
> SIP/2.0 200 OK<br>
> Record-Route:<br>
> <sip:127.0.0.1:5062;lr=on;ftag=708A9F50-17F;did=28c.89;ice_caller=strip;ice_callee=strip;aset=50;rtpprx=yes;vsf=aGFlclVnVW5YdmtjM3pIGi8IUntse2ZsGDkNVg--><br>
> Record-Route:<br>
> <sip:127.0.0.1:6060;r2=on;lr=on;ftag=708A9F50-17F;nat=yes;ngcplb=yes;socket=udp:x.x.x.x:6060><br>
> Record-Route:<br>
> <sip:x.x.x.x:6060;r2=on;lr=on;ftag=708A9F50-17F;nat=yes;ngcplb=yes;socket=udp:x.x.x.x:6060><br>
> Via: SIP/2.0/UDP y.y.y.y:5060;rport=51840;x-route-tag="tgrp:IN"<br>
> From: "WIRELESS CALLER" <sip:2818573448@y.y.y.y>;tag=708A9F50-17F<br>
> To: <sip:<a href="tel:17133751530" value="+17133751530">17133751530</a>@x.x.x.x>;tag=473DB4A8-560465B3000A846D-B0579700<br>
> Call-ID: 5CD39559-2FD111D5-8F95AB03-128F69C7@y.y.y.y<br>
> CSeq: 101 INVITE<br>
> Supported: replaces, path, eventlist<br>
> User-Agent: Grandstream HT701 1.0.7.3<br>
> Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, UPDATE<br>
> Content-Type: application/sdp<br>
> Content-Length: 307<br>
> Contact:<br>
> <sip:ngcp-lb@x.x.x.x:6060;ngcpct=7369703a3132372e302e302e313a353038303b707278726f7574653d31><br>
><br>
> v=0<br>
> o=000B827B5aaa 8000 8000 IN IP4 x.x.x.x<br>
> s=SIP Call<br>
> c=IN IP4 x.x.x.x<br>
> t=0 0<br>
> m=audio 32660 RTP/AVP 18 0 8 101<br>
> a=rtpmap:18 G729/8000<br>
> a=rtpmap:0 PCMU/8000<br>
> a=rtpmap:8 PCMA/8000<br>
> a=rtpmap:101 telephone-event/8000<br>
> a=fmtp:101 0-16,32-36,54<br>
> a=ptime:20<br>
> a=direction:active<br>
> a=sendrecv<br>
> a=rtcp:32661<br>
><br>
> *Apr 14 05:54:00.717: Sent:<br>
> ACK<br>
> sip:x.x.x.x:6060;r2=on;lr=on;ftag=708A9F50-17F;nat=yes;ngcplb=yes;socket=udp:x.x.x.x:6060<br>
> SIP/2.0<br>
> Via: SIP/2.0/UDP y.y.y.y:5060;x-route-tag="tgrp:IN"<br>
> From: "WIRELESS CALLER" <sip:2818573448@y.y.y.y>;tag=708A9F50-17F<br>
> To: <sip:<a href="tel:17133751530" value="+17133751530">17133751530</a>@x.x.x.x>;tag=473DB4A8-560465B3000A846D-B0579700<br>
> Date: Sat, 14 Apr 2001 05:53:52 GMT<br>
> Call-ID: 5CD39559-2FD111D5-8F95AB03-128F69C7@y.y.y.y<br>
> Route:<br>
> <sip:127.0.0.1:6060;r2=on;lr=on;ftag=708A9F50-17F;nat=yes;ngcplb=yes;socket=udp:x.x.x.x:6060>,<sip:127.0.0.1:5062;lr=on;ftag=708A9F50-17F;did=28c.89;ice_caller=strip;ice_callee=strip;aset=50>,<br>
> <sip:ngcp-lb@x.x.x.x:6060;ngcpct=7369703a3132372e302e302e313a353038303b707278726f7574653d31><br>
> Max-Forwards: 6<br>
> Content-Length: 0<br>
> CSeq: 101 ACK<br>
><br>
><br>
> ---<br>
> Best regards,<br>
><br>
</div></div>> *Tung Tran*<br>
><br>
> *V247 Enterprise Corp<br>
> *<a href="tel:713.358.2257" value="+17133582257">713.358.2257</a> office | <a href="tel:281.857.3448" value="+12818573448">281.857.3448</a> cell<br>
> 9999 Bellaire Blvd., Ste. 1111* | *Houston, TX 77036<br>
> *tung.tran@V247.com | <a href="http://www.V247.com" rel="noreferrer" target="_blank">www.V247.com</a> <<a href="http://www.v247.com/" rel="noreferrer" target="_blank">http://www.v247.com/</a>>*<br>
<span class="">><br>
><br>
> On Thu, Sep 24, 2015 at 2:21 PM, Daniel Grotti <<a href="mailto:dgrotti@sipwise.com">dgrotti@sipwise.com</a><br>
</span><span class="">> <mailto:<a href="mailto:dgrotti@sipwise.com">dgrotti@sipwise.com</a>>> wrote:<br>
><br>
> Hi,<br>
> How the ack message looks like?<br>
> Can you paste here the ack from Cisco to SPCE ?<br>
><br>
><br>
> Daniel<br>
><br>
> On Sep 24, 2015 8:36 PM, Tung Tran <<a href="mailto:tung.tran@v247.com">tung.tran@v247.com</a><br>
</span><span class="">> <mailto:<a href="mailto:tung.tran@v247.com">tung.tran@v247.com</a>>> wrote:<br>
> ><br>
> > Dear all<br>
> ><br>
> > I have this scenario:<br>
> > Caller from PSTN (via Cisco gateway 5400) dialed DID number<br>
> which was assigned to an ATA ( grand stream HT701). Call was hit<br>
> SPCE just fine and connected but no audio from both sides, and<br>
> call was disconnect after 30 seconds<br>
> > I ran wireshark on SPCE and saw SPCE didnt forward ACK<br>
> message from Cisco gateway to client (HT701), so I guest HT701<br>
> was waiting for ACK before starting RTP session, and it dropped<br>
> call after timeout<br>
> ><br>
> > Please see the call flow for detail<br>
> ><br>
> ><br>
> ><br>
> > Anyone got that issue before please share how to fix it?<br>
> ><br>
> ><br>
> > ---<br>
> > Best regards,<br>
> ><br>
> > Tung Tran<br>
> ><br>
> > V247 Enterprise Corp<br>
> > <a href="tel:713.358.2257" value="+17133582257">713.358.2257</a> office | <a href="tel:281.857.3448" value="+12818573448">281.857.3448</a> cell<br>
> > 9999 Bellaire Blvd., Ste. 1111 | Houston, TX 77036<br>
</span>> > tung.tran@V247.com | <a href="http://www.V247.com" rel="noreferrer" target="_blank">www.V247.com</a> <<a href="http://www.V247.com" rel="noreferrer" target="_blank">http://www.V247.com</a>><br>
<div class="HOEnZb"><div class="h5">> ><br>
><br>
><br>
><br>
><br>
> _______________________________________________<br>
> Spce-user mailing list<br>
> <a href="mailto:Spce-user@lists.sipwise.com">Spce-user@lists.sipwise.com</a><br>
> <a href="https://lists.sipwise.com/listinfo/spce-user" rel="noreferrer" target="_blank">https://lists.sipwise.com/listinfo/spce-user</a><br>
><br>
<br>
</div></div></blockquote></div><br></div>