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

[Tung Tran]

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> 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> 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
> >
>
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