Well are you sure the scenario you said is working with Xlite is exactly as
you described? It seems wrong to me..... I think a 200Ok and an ACK are
mandatory for the INVITE....
So in this scenario, I think you are missing to expect a 200Ok from
Freeswitch, then send an ACK to it. Only after this you can pause your
script and then send the BYE, which should also expect a 200Ok.
On Mon, Jan 7, 2013 at 1:09 PM, Rajesh Bansal <bansal.rajes...@gmail.com>wrote:
> Hi Rob,
>
> I am using sipp-3.3 at RHEL 5.3 (32 bit).
>
> I want to test following SIP scenario with SIPP. This is produced by
> making a successful SIP call from Xlite softphone client to freeswitch
> (Application Hosted).
>
> CLIENT Freeswtich
>
> Invite ------------>
> <---------- 100 Trying
> <---------- 183 with SDP
> <-- Media Play -->
> Bye ------------>
> <--------------- 200 OK
>
>
> I am getting following
>
> Invite ------------>
> <----------- 100 Trying
> <----------- 183 with SDP
> Cancel ------------>
> <----------- 200 OK
>
> Following is my scenario file.
>
>
> <?xml version="1.0" encoding="ISO-8859-1" ?>
> <!DOCTYPE scenario SYSTEM "sipp.dtd">
>
> <scenario name="UAC with media">
> <send retrans="500">
>
> <![CDATA[
>
> INVITE sip:[service]@[remote_ip]:[remote_port] SIP/2.0
> Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
> From: [field0]
> <sip:[field0]@[local_ip]:[local_port]>;tag=[call_number]
> To: sut <sip:[service]@[remote_ip]:[remote_port]>
> Call-ID: [call_id]
> CSeq: 1 INVITE
> Contact: sip:field0@[local_ip]:[local_port]
> Max-Forwards: 70
> Subject: Performance Test
> Content-Type: application/sdp
> Content-Length: [len]
>
> v=0
> o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
> s=-
> c=IN IP[local_ip_type] [local_ip]
> t=0 0
> m=audio [auto_media_port] RTP/AVP 8
> a=rtpmap:8 PCMA/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-11,16
>
> ]]>
>
> </send>
>
> <recv response="100" optional="true">
> </recv>
>
> <recv response="183">
> </recv>
>
> <pause milliseconds="10000"/>
> <nop>
> <action>
> <exec play_pcap_audio="pcap/mod_01012013_240to21.pcap"/>
> </action>
> </nop>
>
> <send retrans="500">
> <![CDATA[
>
> BYE sip:[service]@[remote_ip]:[remote_port] SIP/2.0
> Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
> From: field0 <sip:field0@[local_ip]:[local_port]>;tag=[call_number]
> To: sut <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]
> Call-ID: [call_id]
> CSeq: 2 BYE
> Contact: sip:field0@[local_ip]:[local_port]
> Max-Forwards: 70
> Subject: Performance Test
> Content-Length: 0
> ]]>
> </send>
>
> <timewait milliseconds="1000"/>
> <ResponseTimeRepartition value="10, 20, 30, 40, 50, 100, 150, 200"/>
> <CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000"/>
> </scenario>
>
> Attached are Files.
>
> 1. Successful SIP Call PCAP sip_succ.pcap
> 2. Failed SIPP call failed_sipp.pcap
> 3. Scenario Used in SIPP crbt1.xml
>
> Best Regards,
> Rajesh Bansal
>
>
> On Mon, Jan 7, 2013 at 8:11 PM, Rob Day <r...@rkd.me.uk> wrote:
>
>> On 7 January 2013 13:31, Rajesh Bansal <bansal.rajes...@gmail.com> wrote:
>> > I need urgent help. I want to test ring back tone application in which
>> we
>> > are playing media after 183(with SDP). But i am not able to make XML
>> (UAC)
>> > scenario for the same. Kindly help me in this regard,
>>
>> Well - what problem are you having? What have you tried so far, what
>> are you expecting it to do, and what is actually happening?
>>
>> (You may find that http://www.catb.org/esr/faqs/smart-questions.html
>> helps you to make the best use of mailing lists such as these,
>> particularly http://www.catb.org/esr/faqs/smart-questions.html#beprecise
>> ).
>>
>>
>> Best,
>> Rob
>>
>
>
>
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