[SR-Users] loose_route() for ACK message

[Laurent Schweizer]

Hello all,

 

I have a script working with openser1.2.1 and I try to move them to kamailio
3.2.1

 

I have actually an issue with the loose_route() function at least for ACK
message.

 

When I get the ACK the loose_route function is changing the URI with the
value of the route header of the ACK message . 

The problem is that this value is the local proxy himself so  the t_relay()
function send the ACK on the loopback interface.

 

Any Idea ?

 

Regards

Laurent

 

Apr 16 18:48:08 proxy kamailio[28033]: INFO: <script>: START REQ:[ACK]
Tran:[7070] URI:[sip:peoplefone at 95.128.80.93:7070]
from:[sip:41763770377 at sip.call.lt;user=phone]
to:[sip:37052077800 at 81.16.238.195;user=phone]
CallID:[25694-UI-05aa645a-7000c6ef6 at sip.call.lt]

Apr 16 18:48:08 proxy kamailio[28033]: INFO: <script>: SANITY OK, REQ:[ACK]
Tran:[7070] URI:[sip:peoplefone at 95.128.80.93:7070]
from:[sip:41763770377 at sip.call.lt;user=phone]
to:[sip:37052077800 at 81.16.238.195;user=phone]
CallID:[25694-UI-05aa645a-7000c6ef6 at sip.call.lt]

Apr 16 18:48:08 proxy kamailio[28033]: INFO: <script>: DIALOG1 REQ:[ACK]
Tran:[7070] URI:[sip:peoplefone at 95.128.80.93:7070]
from:[sip:41763770377 at sip.call.lt;user=phone]
to:[sip:37052077800 at 81.16.238.195;user=phone]
CallID:[25694-UI-05aa645a-7000c6ef6 at sip.call.lt]

If (loose_route()) is called

Apr 16 18:48:08 proxy kamailio[28033]: INFO: <script>: DIALOG1ACK REQ:[ACK]
Tran:[5060] URI:[sip:81.16.238.195;lr;ftag=25694-BR-05aa645b-0f448aeb2]
from:[sip:41763770377 at sip.call.lt;user=phone]
to:[sip:37052077800 at 81.16.238.195;user=phone]
CallID:[25694-UI-05aa645a-7000c6ef6 at sip.call.lt]

Apr 16 18:48:08 proxy kamailio[28033]: INFO: <script>: NATMANAGE REQ:[ACK]
Tran:[5060] URI:[sip:81.16.238.195;lr;ftag=25694-BR-05aa645b-0f448aeb2]
from:[sip:41763770377 at sip.call.lt;user=phone]
to:[sip:37052077800 at 81.16.238.195;user=phone]
CallID:[25694-UI-05aa645a-7000c6ef6 at sip.call.lt]

Apr 16 18:48:08 proxy kamailio[28033]: INFO: <script>: DIAL_RELAY1 REQ:[ACK]
Tran:[5060] URI:[sip:81.16.238.195;lr;ftag=25694-BR-05aa645b-0f448aeb2]
from:[sip:41763770377 at sip.call.lt;user=phone]
to:[sip:37052077800 at 81.16.238.195;user=phone]
CallID:[25694-UI-05aa645a-7000c6ef6 at sip.call.lt]

Apr 16 18:48:08 proxy kamailio[28033]: INFO: <script>: T_RELAY REQ:[ACK]
Tran:[5060] URI:[sip:81.16.238.195;lr;ftag=25694-BR-05aa645b-0f448aeb2]
from:[sip:41763770377 at sip.call.lt;user=phone]
to:[sip:37052077800 at 81.16.238.195;user=phone]
CallID:[25694-UI-05aa645a-7000c6ef6 at sip.call.lt]

 

 

 

U 81.16.238.195:5060 -> 87.110.233.1:5060

SIP/2.0 200 OK.

To: <sip:37052077800 at 81.16.238.195;user=phone>;tag=e868748c.

Via: SIP/2.0/UDP 87.110.233.1:5060;rport=5060;branch=z9hG4bK-26E6-2AB3E9F.

Record-Route: <sip:81.16.238.195;lr;ftag=25694-BR-05aa645b-0f448aeb2>.

CSeq: 79950814 INVITE.

Call-ID: 25694-UI-05aa645a-7000c6ef6 at sip.call.lt.

From:
<sip:41763770377 at sip.call.lt;user=phone>;tag=25694-BR-05aa645b-0f448aeb2.

Content-Type: application/sdp.

Contact: <sip:peoplefone at 95.128.80.93:7070>.

Content-Length: 205.

.

v=0.

o=90706832381 5012 11 IN IP4 81.16.238.195.

s=Mapping.

c=IN IP4 81.16.238.195.

t=0 0.

m=audio 8170 RTP/AVP 8 101.

a=rtpmap:8 PCMA/8000.

a=rtpmap:101 telephone-event/8000.

a=fmtp:101 0-16.

a=sendrecv.

 

 

U 87.110.233.1:5060 -> 81.16.238.195:5060

ACK sip:peoplefone at 95.128.80.93:7070 SIP/2.0.

Call-ID: 25694-UI-05aa645a-7000c6ef6 at sip.call.lt.

Contact: <sip:87.110.233.1:5060>.

CSeq: 79950814 ACK.

From:
<sip:41763770377 at sip.call.lt;user=phone>;tag=25694-BR-05aa645b-0f448aeb2.

Max-Forwards: 30.

Route: <sip:81.16.238.195;lr;ftag=25694-BR-05aa645b-0f448aeb2>.

To: <sip:37052077800 at 81.16.238.195;user=phone>;tag=e868748c.

User-Agent: Cirpack/v4.42o (gw_sip).

Via: SIP/2.0/UDP 87.110.233.1:5060;branch=z9hG4bK-388F-2AB3F03.

Content-Length: 0.

.

 

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