Yep, it's on proxy. I'm attaching the SIPp scenario and the output. The ACK to the 200 OK is not being matched by freeSWITCH.... i've seen too many traces already, looking for some new fresh eyes to help me out.
Regards, David Villasmil email: [email protected] phone: +34669448337 On Tue, Sep 20, 2022 at 8:21 AM Brian West <[email protected]> wrote: > make sure to enable 3pcc in your sofia profiles that are involved. > > /b > > > On Thu, Sep 15, 2022 at 7:01 AM Patrick Karton <[email protected]> > wrote: > >> Check record route header that seems to be the issue. >> >> >> >> Le 15 sept. 2022 12:02, David Villasmil <[email protected]> >> a écrit : >> >> Hello all, >> >> Anyone got a SIPp scenario for reINVITE with late offer? I.e.: >> >> UAT--->INVITE (sdp)--->SIPp >> UAT<--- 100 <---SIPp >> UAT<--- 183 <---SIPp >> UAT<--- 200 OK <---SIPp >> UAT----> ACK --->SIPp >> UAT<--- INVITE (NO sdp) <---SIPp >> UAT---> 100 --->SIPp >> UAT---> 183 --->SIPp >> UAT---> 200 OK (sdp) -->SIPp >> UAT<--- ACK (sdp) <---SIPp >> UAT---> BYE -->SIPp >> UAT<--- ACK <--SIPp >> >> I can’t get past >> >> UAT<--- INVITE (NO sdp) <---SIPp >> >> >> Kamailio responds with “not here” >> >> >> Appreciate the help! >> >> >> >> >> -- >> Regards, >> >> David Villasmil >> email: [email protected] >> phone: +34669448337 >> >> >> _________________________________________________________________________ >> >> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com >> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN >> services. >> Build your next product on our scalable cloud platform. >> >> Join our online community to chat in real time >> https://signalwire.community >> >> Professional FreeSWITCH Services >> [email protected] >> https://freeswitch.com >> >> Official FreeSWITCH Sites >> https://freeswitch.com/oss >> https://freeswitch.org/confluence >> https://cluecon.com >> >> FreeSWITCH-users mailing list >> [email protected] >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> https://freeswitch.com > > > > -- > > Brian West | Co-founder and Developer > > Need Commercial support? email [email protected] > > FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 > <https://maps.google.com/?q=17345+Civic+Drive+%232531+Brookfield,+WI+53045&entry=gmail&source=g> > > Email: [email protected] > > Mobile: 918-424-9378 > > Website: https://www.FreeSWITCH.com <https://www.freeswitch.com/> > > [image: https://www.facebook.com/signalwireinc?src=email] > <https://www.facebook.com/freeswitch> [image: > https://twitter.com/freeswitch] <https://twitter.com/freeswitch> > __________________________________________________________ > Kamailio - Users Mailing List - Non Commercial Discussions > * [email protected] > Important: keep the mailing list in the recipients, do not reply only to > the sender! > Edit mailing list options or unsubscribe: > * https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users >
<?xml version="1.0" encoding="ISO-8859-1"?>
<!DOCTYPE scenario SYSTEM "sipp.dtd">
<scenario name="UAS responder with PCAP/WAV media">
<recv request="OPTIONS" optional="true">
</recv>
<recv crlf="true" rrs="true" request="INVITE" >
<action>
<ereg regexp=".*" search_in="hdr" header="From: " assign_to="1" />
<ereg regexp=".*" search_in="hdr" header="To: " assign_to="2" />
</action>
</recv>
<send>
<![CDATA[
SIP/2.0 180 Ringing
[last_Via:]
[last_From:]
[last_To:];tag=[call_number]
[last_Call-ID:]
[last_CSeq:]
Contact: <sip:[local_ip]:[local_port]>
Content-Length: 0
]]>
</send>
<send retrans="500">
<![CDATA[
SIP/2.0 200 OK
[last_Via:]
[last_Record-Route:]
[last_From:]
[last_To:];tag=[call_number]
[last_Call-ID:]
[last_CSeq:]
Contact: <sip:[local_ip]:[local_port]>
Content-Type: application/sdp
Allow: INVITE, BYE, ACK, OPTIONS, CANCEL, PRACK, UPDATE
Content-Length: [len]
v=0
o=- 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 9
a=rtpmap:9 G722/8000
a=ptime:20
]]>
</send>
<recv request="ACK" />
<!--recv rtd="true" crlf="true" request="ACK" > </recv-->
<!-- send RTP stream from wav file -->
<nop>
<action>
<exec rtp_stream="g722.pcap"/>
</action>
</nop>
<!--recv request="BYE" timeout="5000" ontimeout="2" next="3" /-->
<recv request="BYE" timeout="5000" ontimeout="4" next="3" />
<!-- send INVITE -->
<label id="4"/>
<send>
<![CDATA[
INVITE [next_url] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];rport;branch=[branch]
Route: <sip:172.18.2.142:5088;lr>
From: [$2];tag=[call_number]
To: [$1]
[last_Call-ID:]
Cseq: 3 INVITE
Contact: <sip:[local_ip]:[local_port]>
Max-Forwards: 70
User-Agent: SIPP
Subject: Performance Test
Content-Type: application/sdp
Allow: INVITE, BYE, ACK, OPTIONS, CANCEL, PRACK, UPDATE
Content-Length: [len]
]]>
</send>
<!-- receive TRYING -->
<recv response="100" optional="true">
</recv>
<recv response="200">
<action>
<ereg regexp=".*"
search_in="hdr"
header="Contact: "
assign_to="3" />
<log message="Contact: <[$3]>"/>
</action>
</recv>
<!-- send ACK -->
<send>
<![CDATA[
ACK [next_url] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];rport;branch=[branch]
Route: <sip:172.18.2.142:5088;lr>
From: [$2];tag=[call_number]
To: [$1]
Contact: <sip:[local_ip]:[local_port]>
[last_Call-ID:]
CSeq: [cseq+1] ACK
Max-Forwards: 70
Content-Length: [len]
Content-Type: application/sdp
v=0
o=- 53655765 2353687637 IN IP[local_ip_type] [local_ip]
s=-
c=IN IP[media_ip_type] [media_ip]
t=0 0
m=audio 10468 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=rtcp:10469
a=ptime:20
]]>
</send>
<nop>
<action>
<exec rtp_stream="g711a.pcap"/>
</action>
</nop>
<!--recv request="BYE" timeout="5000" ontimeout="2" next="3" /-->
<label id ="5"/>
<recv request="BYE" timeout="1890000" ontimeout="2" next="3" />
<!-- send BYE after timeout set avove (the audio file is 1890000 miliseconds long) -->
<label id="2"/>
<send retrans="500">
<![CDATA[
BYE [next_url] SIP/2.0
[last_Via:]
[last_Record-Route:]
Route: <sip:172.18.2.142:5088;lr>
From: [$2];tag=[call_number]
To: [$1]
[last_Call-ID:]
CSeq: [cseq] BYE
Contact: <sip:[local_ip]:[local_port]>
Max-Forwards: 70
User-Agent: SIPP
Subject: Performance Test
Content-Length: 0
]]>
</send>
<!-- receive OK -->
<recv response="200" next="99"/>
<!-- Send an OK if we receive a BYE -->
<label id="3"/>
<send>
<![CDATA[
SIP/2.0 200 OK
[last_Via:]
[last_From:]
[last_To:]
[last_Call-ID:]
[last_CSeq:]
Contact: <sip:[local_ip]:[local_port]>
Content-Length: 0
]]>
</send>
<label id="99"/>
<!-- Keep the call open for a while in case the 200 is lost to be -->
<!-- able to retransmit it if we receive the BYE again. -->
<pause milliseconds="4000"/>
<!-- 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>
message_file.log
Description: Binary data
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