Hello Pavel, Could you please review my script an suggest the required changes.
modified_reg_with_call.xml > <?xml version="1.0" encoding="ISO-8859-1" ?> > <!DOCTYPE scenario SYSTEM "sipp.dtd"> > <scenario name="REGISTRATION with INCOMING CALL"> > <send retrans="500"> > <![CDATA[ > REGISTER sip:[field1];transport=tls SIP/2.0 > Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch] > From: <sip:[field0]@[field1]>;tag=[call_number] > To: <sip:[field0]@[field1]:[remote_port]> > Call-ID: [call_id] > CSeq: 1 REGISTER > Contact: sip:[field0]@[local_ip]:[local_port] > Max-Forwards: 5 > Expires: 3600 > Allow: > INVITE,ACK,BYE,CANCEL,OPTIONS,MESSAGE,INFO,UPDATE,REGISTER,REFER,NOTIFY > k: timer,path,replaces > User-Agent: SIPp > Content-Length: 0 > ]]> > </send> > > <recv response="401" auth="true"></recv> > > <send retrans="500"> > <![CDATA[ > REGISTER sip:[field1];transport=tls SIP/2.0 > Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch] > From: <sip:[field0]@[field1]>;tag=[call_number] > To: <sip:[field0]@[field1]> > Call-ID: [call_id] > CSeq: 2 REGISTER > Contact: sip:[field0]@[local_ip]:[local_port] > [field2] > Max-Forwards: 5 > Expires: 3600 > Allow: > INVITE,ACK,BYE,CANCEL,OPTIONS,MESSAGE,INFO,UPDATE,REGISTER,REFER,NOTIFY > k: timer,path,replaces > User-Agent: SIPp > Content-Length: 0 > ]]> > </send> > > <recv response="100" optional="true"></recv> > <recv response="200"></recv> > > <recv request="INVITE" next="handle_INVITE"/> > <label id="handle_INVITE"/> > > <send> > <![CDATA[ > SIP/2.0 100 Trying > Record-Route: [$2] > Record-Route: [$3] > Via: [$4] > Via: [$5] > [last_From:] > [last_To:] > [last_Call-ID:] > [last_CSeq:] > Contact: <sip:[local_ip]:[local_port];transport=[transport]> > Content-Length: 0 > ]]> > </send> > > <send> > <![CDATA[ > SIP/2.0 180 Ringing > Record-Route: [$2] > Record-Route: [$3] > Via: [$4] > Via: [$5] > [last_From:] > [last_To:];tag=[call_number] > [last_Call-ID:] > [last_CSeq:] > Contact: <sip:[local_ip]:[local_port];transport=[transport]> > Content-Length: 0 > ]]> > </send> > > <send retrans="500"> > <![CDATA[ > SIP/2.0 200 OK > Record-Route: [$2] > Record-Route: [$3] > Via: [$4] > Via: [$5] > [last_From:] > [last_To:];tag=[call_number] > [last_Call-ID:] > [last_CSeq:] > Contact: <sip:[local_ip]:[local_port];transport=[transport]> > Content-Type: application/sdp > Content-Length: [len] > > v=0 > o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip] > s= Kumartest > c=IN IP[media_ip_type] [media_ip] > t=0 0 > m=audio [media_port] RTP/AVP 0 101 13 > a=rtpmap:0 PCMU/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-15 > a=rtpmap:13 CN/8000 > ]]> > </send> > > <recv request="ACK" rtd="true" crlf="true"></recv> > > <nop> > <action> > <exec play_pcap_audio="playspeech.pcap"/> > </action> > </nop> > > <recv request="BYE"></recv> > > <send> > <![CDATA[ > SIP/2.0 200 OK > [last_Via:] > [last_From:] > [last_To:] > [last_Call-ID:] > [last_CSeq:] > Contact: <sip:[local_ip]:[local_port];transport=[transport]> > 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 Mon, Dec 9, 2024 at 8:18 PM Šindelka Pavel <sinde...@ttc.cz> wrote: > Hello KJM, > > I'd suggest you to read > https://sourceforge.net/p/sipp/mailman/message/34707334/ as it covers > most of your issue. > > The use of TLS may complicate things further, though - I hazily remember > that I had to patch something in the source code so that sipp could > initiate outgoing TCP sessions and also expect incoming ones. But on the > other hand, since you start by sending a REGISTER, the typical use case > would be that the remote party would not create a new TCP session to > deliver the INVITE to your scenario but rather reuse the existing one > created by the REGISTER, so that patch may not be necessary. > > Pavel > > > Dne 09.12.2024 v 14:48 Jyotirmaya Mohanty napsal(a): > > Hello Team, > > I have a query regarding SIPp script. want to REGISTER and Receive an > incoming call on SIPp over TLS. Is it possible? > > Messages Retrans Timeout > Unexpected-Msg > REGISTER ----------> 1 0 0 > 401 <---------- 1 0 0 0 > REGISTER ----------> 1 0 0 > 100 <---------- 1 0 0 0 > 200 <---------- 1 0 0 0 > INVITE <---------- 0 0 0 0 > > 100 ----------> 0 0 > 180 ----------> 0 0 > 200 ----------> 0 0 0 > ACK <---------- E-RTD1 0 0 0 0 > > [ NOP ] > BYE <---------- 0 0 0 0 > 200 ----------> 0 0 > ------- Waiting for active calls to end. Press [q] again to force exit. > ------- > > > My Registration is successful, but while receiving INVITE I am getting > 2024-12-09 13:18:31.970086 1733750311.970086: Discarding message > which can't be mapped to a known SIPp call: > INVITE sip:user_KvwJyyyKuu@88.112.45.95:5069 > <http://sip:user_KvwJyyyKuu@38.102.145.195:5069/> SIP/2.0 > > For UDP transport and if I ran separate script for REGISTER and INVITE > UAS. It is working fine. > > Please let me know and Thanks in advance, how to run it for TLS > > Best Regards > KJM > > > _______________________________________________ > Sipp-users mailing > listSipp-users@lists.sourceforge.nethttps://lists.sourceforge.net/lists/listinfo/sipp-users > > -- > > > *Ing. Pavel Šindelka * senior specialista > > > > > > TTC MARCONI s. r. o. > Třebohostická 987/5, 100 00 Praha 10 > +420 234 051 712, +420 602 355 199 > sinde...@ttc.cz, www.ttc-marconi.com > > _______________________________________________ > Sipp-users mailing list > Sipp-users@lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/sipp-users >
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