For anyone who might ever need it, i'm attaching a working version.
Regards,
David Villasmil
email: [email protected]
phone: +34669448337
On Tue, Sep 20, 2022 at 10:36 AM David Villasmil <
[email protected]> wrote:
> 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
>>>
>>>
>>> _________________________________________________________________________
>>>
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>>
>>
>>
>> --
>>
>> 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
>>
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>>
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>> __________________________________________________________
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>> * [email protected]
>> Important: keep the mailing list in the recipients, do not reply only to
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>>
>
<?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: [cseq+1] 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:172.18.2.142:5060;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 2353687638 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>
__________________________________________________________
Kamailio - Users Mailing List - Non Commercial Discussions
* [email protected]
Important: keep the mailing list in the recipients, do not reply only to the
sender!
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