On 23 Dec 2024, at 7:33 PM, Alex Balashov via sr-users
<sr-users@lists.kamailio.org<mailto:sr-users@lists.kamailio.org>> wrote:
Why is the Via branch on the ACK different to that of the INVITE or the response?
See RFC 3261 § 17.1.1.3 ("Construction of the ACK Request"):
"The ACK MUST contain a single Via header field, and
this MUST be equal to the top Via header field of the original
request."
You have:
INVITE sip:123456@192.168.86.128:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.86.250:5060;branch=z9hG4bK-35646-1-0
SIP/2.0 302 Redirect
Via: SIP/2.0/UDP 192.168.86.250:5060;branch=z9hG4bK-35646-1-0
ACK sip:123456@192.168.86.128:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.86.250:5060;branch=z9hG4bK-35646-1-3
So, it's not being matched.
-- Alex
On Dec 23, 2024, at 5:14 pm, Alexis Fidalgo via
sr-users <sr-users@lists.kamailio.org<mailto:sr-users@lists.kamailio.org>>
wrote:
request_route:
if(is_method("ACK")){
if(!t_check_trans()){
xlog("L_INFO","AAAA: MATCHED TRANSACTION\n"); }
else {
xlog("L_INFO","BBBB: MATCHED TRANSACTION\n");
}
exit;
}
if(is_method("INVITE")){
t_newtran();
http_async_query("http://nuc:8080<http://nuc:8080/>",
"HTTP_REPLY");
}
route[HTTP_REPLY] {
if ($http_ok) {
xlog("L_INFO", "route[HTTP_REPLY]: status $http_rs\n");
xlog("L_INFO", "route[HTTP_REPLY]: body $http_rb\n");
append_branch("sip:1111@otherdomain.com:5060");
t_reply(302,"Redirect");
} else {
xlog("L_INFO", "route[HTTP_REPLY]: error $http_err)\n");
}
}
pcap list
15 173.459633617 192.168.86.250 → 192.168.86.128 SIP/SDP 588 Request: INVITE
sip:123456@192.168.86.128:5060 |
16 173.460549905 192.168.86.128 → 192.168.86.250 SIP 344 Status: 100 Trying |
17 173.762804083 192.168.86.128 → 192.168.86.250 SIP 467 Status: 302 Redirect |
18 173.789493070 192.168.86.250 → 192.168.86.128 SIP 415 Request: ACK
sip:123456@192.168.86.128:5060 |
19 174.213115954 192.168.86.128 → 192.168.86.250 SIP 467 Status: 302 Redirect |
20 175.213132963 192.168.86.128 → 192.168.86.250 SIP 467 Status: 302 Redirect |
21 177.213136064 192.168.86.128 → 192.168.86.250 SIP 467 Status: 302 Redirect |
sipp trace dump
----------------------------------------------- 2024-12-23 19:10:01.820622
UDP message sent (546 bytes):
INVITE sip:123456@192.168.86.128:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.86.250:5060;branch=z9hG4bK-35646-1-0
From: sipp <sip:sipp@192.168.86.250:5060>;tag=35646SIPpTag001
To: 123456 <sip:123456@192.168.86.128:5060>
Call-ID: 1-35646@192.168.86.250<mailto:1-35646@192.168.86.250>
CSeq: 1 INVITE
Contact: sip:sipp@192.168.86.250:5060
Max-Forwards: 70
Subject: Performance Test
Content-Type: application/sdp
Content-Length: 139
v=0
o=user1 53655765 2353687637 IN IP4 192.168.86.250
s=-
c=IN IP4 192.168.86.250
t=0 0
m=audio 6000 RTP/AVP 0
a=rtpmap:0 PCMU/8000
----------------------------------------------- 2024-12-23 19:10:01.847444
UDP message received [302] bytes :
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.86.250:5060;branch=z9hG4bK-35646-1-0
From: sipp <sip:sipp@192.168.86.250:5060>;tag=35646SIPpTag001
To: 123456 <sip:123456@192.168.86.128:5060>
Call-ID: 1-35646@192.168.86.250<mailto:1-35646@192.168.86.250>
CSeq: 1 INVITE
Server: kamailio (5.8.4 (x86_64/linux))
Content-Length: 0
----------------------------------------------- 2024-12-23 19:10:02.169378
UDP message received [425] bytes :
SIP/2.0 302 Redirect
Via: SIP/2.0/UDP 192.168.86.250:5060;branch=z9hG4bK-35646-1-0
From: sipp <sip:sipp@192.168.86.250:5060>;tag=35646SIPpTag001
To: 123456
<sip:123456@192.168.86.128:5060>;tag=a6a1c5f60faecf035a1ae5b6e96e979a-e0f89f75
Call-ID: 1-35646@192.168.86.250<mailto:1-35646@192.168.86.250>
CSeq: 1 INVITE
Contact: <sip:1111@otherdomain.com:5060>, <sip:1111@otherdomain.com:5060>
Server: kamailio (5.8.4 (x86_64/linux))
Content-Length: 0
----------------------------------------------- 2024-12-23 19:10:02.169921
UDP message sent (373 bytes):
ACK sip:123456@192.168.86.128:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.86.250:5060;branch=z9hG4bK-35646-1-3
From: sipp <sip:sipp@192.168.86.250:5060>;tag=35646SIPpTag001
To: 123456
<sip:123456@192.168.86.128:5060>;tag=a6a1c5f60faecf035a1ae5b6e96e979a-e0f89f75
Call-ID: 1-35646@192.168.86.250<mailto:1-35646@192.168.86.250>
CSeq: 1 ACK
Max-Forwards: 70
Subject: Performance Test
Content-Length: 0
sipp br.xml
<?xml version="1.0" encoding="ISO-8859-1" ?>
<!DOCTYPE scenario SYSTEM "sipp.dtd">
<scenario name="Basic Sipstone UAC">
<!-- In client mode (sipp placing calls), the Call-ID MUST be -->
<!-- generated by sipp. To do so, use [call_id] keyword. -->
<send retrans="500">
<![CDATA[
INVITE sip:[service]@[remote_ip]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: sipp
<sip:sipp@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]
To: [service] <sip:[service]@[remote_ip]:[remote_port]>
Call-ID: [call_id]
CSeq: 1 INVITE
Contact: sip:sipp@[local_ip]:[local_port]
Max-Forwards: 70
Subject: Performance Test
Content-Type: application/sdp
Content-Length: [len]
v=0
o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
s=-
c=IN IP[media_ip_type] [media_ip]
t=0 0
m=audio [media_port] RTP/AVP 0
a=rtpmap:0 PCMU/8000
]]>
</send>
<recv response="100" optional="true"></recv>
<!-- By adding rrs="true" (Record Route Sets), the route sets
-->
<!-- are saved and used for following messages sent. Useful to test -->
<!-- against stateful SIP proxies/B2BUAs. -->
<recv response="302" rtd="true">
</recv>
<!-- Packet lost can be simulated in any send/recv message by -->
<!-- by adding the 'lost = "10"'. Value can be [1-100] percent.
-->
<send>
<![CDATA[
ACK sip:[service]@[remote_ip]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: sipp
<sip:sipp@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]
To: [service] <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]
Call-ID: [call_id]
CSeq: 1 ACK
Max-Forwards: 70
Subject: Performance Test
Content-Length: 0
]]>
</send>
<!-- definition of the response time repartition table (unit is ms) -->
<ResponseTimeRepartition value="10, 20, 30, 40, 50, 100, 150, 200"/>
<!-- definition of the call length repartition table (unit is ms) -->
<CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000"/>
</scenario>
On 23 Dec 2024, at 6:44 PM, Ben Kaufman
<bkaufman@bcmone.com<mailto:bkaufman@bcmone.com>> wrote:
Does your 3xx message have a Contact, and does that Contact URI match the RURI in your
ACK?
Kaufman
Senior Voice Engineer
E: bkaufman@bcmone.com<mailto:bkaufman@bcmone.com>
SIP.US Client Support: 800.566.9810 | SIPTRUNK Client Support: 800.250.6510 |
Flowroute Client Support: 855.356.9768
From: Alexis Fidalgo via sr-users
<sr-users@lists.kamailio.org<mailto:sr-users@lists.kamailio.org>>
Sent: Monday, December 23, 2024 3:19 PM
To: Kamailio (SER) - Users Mailing List
<sr-users@lists.kamailio.org<mailto:sr-users@lists.kamailio.org>>
Cc: Alexis Fidalgo <alzrck@gmail.com<mailto:alzrck@gmail.com>>
Subject: [SR-Users] help on how to get ACK
CAUTION: This email originated from outside the organization. Do not click links or open
attachments unless you recognize the sender and know the content is safe.
Hello all, moving just a bit aside of the http and async_http.
After all the real useful and interesting thread on that topic what helped me, im facing
a problem i cant deal with and need a hint at least.
Scenario
INVITE -> Kamailio
on request_route
...
if(is_method("INVITE")){
t_newtran();
http_async_query("http://nuc:8080<http://nuc:8080/>",
"HTTP_REPLY");
}
…
Kamailio -> 100 - Trying
then
route[HTTP_REPLY] {
if ($http_ok) {
xlog("L_INFO", "route[HTTP_REPLY]: status $http_rs\n");
xlog("L_INFO", "route[HTTP_REPLY]: body $http_rb\n");
t_reply(302,"Redirect");
} else {
xlog("L_INFO", "route[HTTP_REPLY]: error $http_err)\n");
}
}
Kamailio -> 302 Redirect
ACK -> Kamailio
This last ACK, how can i read it and use it to terminate the transaction? because
Kamailio keeps transmitting the 302 message 3 more times until the transaction is finished
by a timer
42(44) DEBUG: tm [t_reply.c:1723]: t_retransmit_reply(): reply retransmitted.
buf=0x7f4c44f9d680: SIP/2.0 3..., shmem=0x7f4c3fce7900: SIP/2.0 3
42(44) DEBUG: tm [t_reply.c:1723]: t_retransmit_reply(): reply retransmitted.
buf=0x7f4c44f9d680: SIP/2.0 3..., shmem=0x7f4c3fce7900: SIP/2.0 3
42(44) DEBUG: tm [t_reply.c:1723]: t_retransmit_reply(): reply retransmitted.
buf=0x7f4c44f9d680: SIP/2.0 3..., shmem=0x7f4c3fce7900: SIP/2.0 3
42(44) DEBUG: tm [timer.c:642]: wait_handler(): finished transaction: 0x7f4c3fcd35a0
(p:0x7f4c3fad85d0/n:0x7f4c3fad85d0)
42(44) DEBUG: tm [h_table.c:133]: free_cell_helper(): freeing transaction 0x7f4c3fcd35a0
from timer.c:651
in request_route i have
if(is_method("ACK")){
if(!t_check_trans()){
t_release();
}
exit;
}
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