Hi Sangdrax,

As you've found, the situation where you need to register before
receiving a call isn't well-supported in SIPp, particularly not for
connection-oriented transports like TCP. I've just replied to a
separate email thread discussing how out-of-call scenarios can help
here (and what their limitations are, which also apply to the very
similar rx scenarios created by the patch you mention) - if you
haven't received it you can find it at
http://sourceforge.net/mailarchive/forum.php?thread_name=CAH1RVig_qj5OSnaKfRkAJ0pY8bOw8idFi3wo9DUZ2H6OAC0NMQ%40mail.gmail.com&forum_name=sipp-users.

Best,
Rob

On 13 January 2014 17:34, sangdrax8 <sangdr...@gmail.com> wrote:
> So you are saying that yours works with this file?  My server requires
> authentication, and isn't using UDP.  Besides adding authentication and
> using TLS, your scenario looks just like mine.  When I start the session as
> a client, it will not recognize the INVITE.
>
> I have found a post talking about a patch that was made to allow a hybrid
> mode, where you can specify an XML file for the client and one to be run
> when receiving connections that don't match.  It says this patch fixes
> multiple things, and was built on version 3.2.  It would appear that what I
> am doing is what this was written for.
>
> Any know if this patch would still work for the 3.99 version, or if these
> features were accepted and rolled into the 3.99 version?  I am guessing they
> were not since I see no documentation of it.
>
> Referenced post
>
> http://permalink.gmane.org/gmane.comp.telephony.sipp.user/5769
>
>
>
>
>
>
> On Fri, Jan 10, 2014 at 11:37 PM, Sakharam Thorat
> <sakharam.tho...@einfochips.com> wrote:
>>
>>
>> Try Following, It should work for user case
>>
>> command -> sipp -sn uac <server ip> -sf scenario.xml -m 1
>>
>> I registered user 2000 to registrar with no password, in your case change
>> scenario accordingly .
>>
>> <?xml version="1.0" encoding="ISO-8859-1" ?>
>> <scenario name="UAC Register">
>>
>>     <send retrans="500">
>>         <![CDATA[
>>
>>         REGISTER sip:2000@[remote_ip]:[remote_port] SIP/2.0
>>
>>         Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
>>         From: v4-in
>> <sip:v4-in@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]
>>         To: <sip:v4-in@[remote_ip]:[remote_port]>
>>         Call-ID: [call_id]
>>         CSeq: 1 REGISTER
>>         Contact: sip:v4-in@[local_ip]:[local_port]
>>         Max-Forwards: 70
>>         Subject: REGISTER Test
>>         Expires: 3600
>>         Content-Length: 0
>>
>>         ]]>
>>     </send>
>>
>>     <recv response="200" rtd="true" />
>>
>>   <recv request="INVITE" crlf="true">
>>   </recv>
>>
>>   <!-- The '[last_*]' keyword is replaced automatically by the
>> -->
>>   <!-- specified header if it was present in the last message received
>> -->
>>   <!-- (except if it was a retransmission). If the header was not
>> -->
>>   <!-- present or if no message has been received, the '[last_*]'
>> -->
>>   <!-- keyword is discarded, and all bytes until the end of the line
>> -->
>>   <!-- are also discarded.
>> -->
>>   <!--
>> -->
>>   <!-- If the specified header was present several times in the
>> -->
>>   <!-- message, all occurences are concatenated (CRLF seperated)
>> -->
>>   <!-- to be used in place of the '[last_*]' keyword.
>> -->
>>
>>   <send>
>>     <![CDATA[
>>
>>       SIP/2.0 180 Ringing
>>       [last_Via:]
>>       [last_From:]
>>       [last_To:];tag=[pid]SIPpTag01[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
>>       [last_Via:]
>>       [last_From:]
>>       [last_To:];tag=[pid]SIPpTag01[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=-
>>       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 request="ACK"
>>         optional="true"
>>         rtd="true"
>>         crlf="true">
>>   </recv>
>>
>>   <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>
>>
>>   <!-- 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.
>> -->
>>   <timewait 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>
>>
>>
>> Best Regards,
>> Sakharam Thorat.
>>
>>
>> ________________________________
>> Date: Fri, 10 Jan 2014 13:37:34 -0500
>> From: sangdr...@gmail.com
>> To: sipp-users@lists.sourceforge.net
>> Subject: [Sipp-users] UAC receive INVITE problems
>>
>>
>> I am trying to have a UAC xml script that can log into my server, then
>> recieve an INVITE and respond to it.  Currently I can register, and even
>> place calls just fine.  I seem to be having issues with the recieving of the
>> invite.  Sipp see's the invite and says it can't be mapped to a known sipp
>> call, even though my scenario is sitting and waiting for an INVITE.  Can
>> someone help me understand how to use the recv?
>>
>> Currently I do the following, and it never considers the INVITE as part of
>> my call so I get stuck
>>
>>
>> <scenario name="User Register">
>> <send>
>>     <![CDATA[
>>         REGISTER sip:[remote_ip] 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: [cseq] REGISTER
>>         Contact: <sip:[field0]@[local_ip]:[local_port];transport=tls>
>>         Max-Forwards: 10
>>         Expires: 120
>>         User-Agent: SIPp
>>         Content-Length: 0
>>     ]]>
>> </send>
>> <recv response="401" auth="true"></recv>
>> <send>
>>     <![CDATA[
>>         REGISTER sip:[remote_ip] 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: [cseq] REGISTER
>>         Contact: <sip:[field0]@[local_ip]:[local_port];transport=tls>
>>         [field2]
>>         Max-Forwards: 10
>>         Expires: 120
>>         User-Agent: SIPp
>>         Content-Length: 0
>>     ]]>
>> </send>
>> <recv response="200"></recv>
>>
>> <recv request="INVITE"></recv>
>>
>>
>>
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