[Spce-user] Missing ACK when dial from peering to SPCE

Fri Sep 25 06:33:52 EDT 2015

Hi Tung,
you may try to enable the parameter
strict_routing_safe: 'yes'
in /etc/ngcp-config/config.yml and do ngcpcfg apply..
The system will do its best to try to route the stict-routed packets.

Regards,
Andrew

On 09/25/2015 08:09 AM, Daniel Grotti wrote:
> Looks like he's doing strict routing an not loose routing. He's setting
> the route value as ruri, and not the contact in 200ok.
> It is not following rfc3261.
> 
> Daniel
> 
> On Sep 24, 2015 11:55 PM, Tung Tran <tung.tran at v247.com> wrote:
> 
>     Hi Daniel
> 
>     Here is 200 OK and ACK from Cisco gateway
> 
>     x.x.x.x: SPCE
>     y.y.y.y: Cisco GW
> 
> 
>     Apr 14 05:54:00.713: Received: 
>     SIP/2.0 200 OK
>     Record-Route:
>     <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-->
>     Record-Route:
>     <sip:127.0.0.1:6060;r2=on;lr=on;ftag=708A9F50-17F;nat=yes;ngcplb=yes;socket=udp:x.x.x.x:6060>
>     Record-Route:
>     <sip:x.x.x.x:6060;r2=on;lr=on;ftag=708A9F50-17F;nat=yes;ngcplb=yes;socket=udp:x.x.x.x:6060>
>     Via: SIP/2.0/UDP  y.y.y.y:5060;rport=51840;x-route-tag="tgrp:IN"
>     From: "WIRELESS CALLER" <sip:2818573448 at y.y.y.y>;tag=708A9F50-17F
>     To: <sip:17133751530 at x.x.x.x>;tag=473DB4A8-560465B3000A846D-B0579700
>     Call-ID: 5CD39559-2FD111D5-8F95AB03-128F69C7 at y.y.y.y
>     CSeq: 101 INVITE
>     Supported: replaces, path, eventlist
>     User-Agent: Grandstream HT701 1.0.7.3
>     Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, UPDATE
>     Content-Type: application/sdp
>     Content-Length: 307
>     Contact:
>     <sip:ngcp-lb at x.x.x.x:6060;ngcpct=7369703a3132372e302e302e313a353038303b707278726f7574653d31>
> 
>     v=0
>     o=000B827B5aaa 8000 8000 IN IP4 x.x.x.x
>     s=SIP Call
>     c=IN IP4 x.x.x.x
>     t=0 0
>     m=audio 32660 RTP/AVP 18 0 8 101
>     a=rtpmap:18 G729/8000
>     a=rtpmap:0 PCMU/8000
>     a=rtpmap:8 PCMA/8000
>     a=rtpmap:101 telephone-event/8000
>     a=fmtp:101 0-16,32-36,54
>     a=ptime:20
>     a=direction:active
>     a=sendrecv
>     a=rtcp:32661
> 
>     *Apr 14 05:54:00.717: Sent: 
>     ACK
>     sip:x.x.x.x:6060;r2=on;lr=on;ftag=708A9F50-17F;nat=yes;ngcplb=yes;socket=udp:x.x.x.x:6060
>     SIP/2.0
>     Via: SIP/2.0/UDP  y.y.y.y:5060;x-route-tag="tgrp:IN"
>     From: "WIRELESS CALLER" <sip:2818573448 at y.y.y.y>;tag=708A9F50-17F
>     To: <sip:17133751530 at x.x.x.x>;tag=473DB4A8-560465B3000A846D-B0579700
>     Date: Sat, 14 Apr 2001 05:53:52 GMT
>     Call-ID: 5CD39559-2FD111D5-8F95AB03-128F69C7 at y.y.y.y
>     Route:
>     <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>,
>     <sip:ngcp-lb at x.x.x.x:6060;ngcpct=7369703a3132372e302e302e313a353038303b707278726f7574653d31>
>     Max-Forwards: 6
>     Content-Length: 0
>     CSeq: 101 ACK
> 
> 
>     ---
>     Best regards,
> 
>     *Tung Tran*
> 
>     *V247 Enterprise Corp
>     *713.358.2257 office   |  281.857.3448 cell
>     9999 Bellaire Blvd., Ste. 1111* | *Houston, TX 77036
>     *tung.tran at V247.com | www.V247.com <http://www.v247.com/>*
> 
> 
>     On Thu, Sep 24, 2015 at 2:21 PM, Daniel Grotti <dgrotti at sipwise.com
>     <mailto:dgrotti at sipwise.com>> wrote:
> 
>         Hi,
>         How the ack message looks like?
>         Can you paste here the ack from Cisco to SPCE ?
> 
> 
>         Daniel
> 
>         On Sep 24, 2015 8:36 PM, Tung Tran <tung.tran at v247.com
>         <mailto:tung.tran at v247.com>> wrote:
>         >
>         > Dear all
>         >
>         > I have this scenario:
>         > Caller from PSTN (via Cisco gateway 5400) dialed DID number
>         which was assigned to an ATA ( grand stream HT701). Call was hit
>         SPCE just fine and connected but no audio from both sides, and
>         call was disconnect after 30 seconds
>         >  I ran wireshark on SPCE and saw SPCE didnt forward ACK
>         message from Cisco gateway to client (HT701), so I guest HT701
>         was waiting for ACK before starting RTP session, and it dropped
>         call after timeout
>         >
>         > Please see the call flow for detail 
>         >
>         >
>         >
>         > Anyone got that issue before please share how to fix it?
>         >
>         >
>         > ---
>         > Best regards,
>         >
>         > Tung Tran
>         >
>         > V247 Enterprise Corp
>         > 713.358.2257 office   |  281.857.3448 cell
>         > 9999 Bellaire Blvd., Ste. 1111 | Houston, TX 77036
>         > tung.tran at V247.com | www.V247.com <http://www.V247.com>
>         >
> 
> 
> 
> 
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