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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