[Spce-user] Missing ACK when dial from peering to SPCE
Tung Tran
tung.tran at v247.com
Fri Sep 25 13:45:03 EDT 2015
Hi Andrew and All
I changed strict_routing_safe: 'yes' and applied but it does't solve my
problem, the lb still drops ACK package from Cisco gateway
Here is the log from kamailio-lb.log
*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 at y.y.y.y T=sip:17133751530 at x.x.x.x IP=udp:y.y.y.y:56266
ID=C476CBE2-307B11D5-9770AB03-128F69C7 at y.y.y.y UA='<null>'*
*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 at y.y.y.y UA='<null>'*
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?
Thank you all
---
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 Fri, Sep 25, 2015 at 5:33 AM, Andrew Pogrebennyk <
apogrebennyk at sipwise.com> wrote:
> 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>
> > >
> >
> >
> >
> >
> > _______________________________________________
> > Spce-user mailing list
> > Spce-user at lists.sipwise.com
> > https://lists.sipwise.com/listinfo/spce-user
> >
>
>
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