Would you mind running the scenario again with the CLI option " -trace_msg "
and providing the log?
In your call-flow, what happens when you replace 'sipp' with another instance
of your softphone?
Eg. Softphone -> proxy -> softphone
JF
From: manaswini.m...@gmail.com [mailto:manaswini.m...@gmail.com]
Sent: Friday, 9 September 2011 5:43 PM
To: sipp-users@lists.sourceforge.net
Subject: [Sipp-users] getting 407 for invite
Hi
i am executing my test case using sipp, the flow will as follows
sipp -> proxy -> softphone
i am sending Invite from my script and getting 407. when second time i send
invite with [authentication username= password=]
i am geting 407 again. i expect there is some error in my script, but i am
unable to find it out.
My script is as belows
<send retrans="500">
<![CDATA[
INVITE sip:bob@[remote_ip]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: "alice" <sip:alice@[local_ip]:[local_port]>;tag=[pid][call_number]
To: "bob" <sip:bob@[remote_ip]:[remote_port]>
Call-ID: [call_id]
CSeq: 1 INVITE
Contact: sip:alice@[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[media_ip_type] [media_ip]
t=0 0
m=audio [media_port] RTP/AVP 0
a=rtpmap:0 PCMU/8000
]]>
</send>
<recv response="407" auth="true" >
</recv>
<send>
<![CDATA[
ACK sip:bob@[remote_ip]:[remote_port] SIP/2.0
[last_Via:]
[last_From:]
[last_To:]
Call-ID: [call_id]
CSeq: 1 ACK
Contact: sip:alice@[local_ip]:[local_port]
Max-Forwards: 70
Subject: Performance Test
Content-Length: 0
]]>
</send>
<send retrans="500">
<![CDATA[
INVITE sip:bob@[remote_ip]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: "alice" <sip:alice@[local_ip]:[local_port]>;tag=[pid][call_number]
To: "bob" <sip:bob@[remote_ip]:[remote_port]
Call-ID: [call_id]
CSeq: 2 INVITE
[authentication username=alice password=alice]
Contact: sip:alice@[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[media_ip_type] [media_ip]
t=0 0
m=audio [media_port] RTP/AVP 0
a=rtpmap:0 PCMU/8000
]]>
</send>
<!-- By adding rrs="true" (Record Route Sets), the route sets -->
<!-- are saved and used for following messages sent. Useful to test -->
<!-- against stateful SIP proxies/B2BUAs. -->
<recv response="100" optional="true">
</recv>
<recv response="180" optional="true">
</recv>
<recv response="200" crlf="true">
</recv>
<!-- Packet lost can be simulated in any send/recv message by -->
<!-- by adding the 'lost = "10"'. Value can be [1-100] percent. -->
<send>
<![CDATA[
ACK sip:bob@[remote_ip]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: "alice" <sip:alice@[local_ip]:[local_port]>;tag=[pid][call_number]
To: bob<sip:bob@[remote_ip]:[remote_port]>[peer_tag_param]
Call-ID: [call_id]
CSeq: 2 ACK
Contact: sip:alice@[local_ip]:[local_port]
Max-Forwards: 70
Subject: Performance Test
Content-Length: 0
]]>
</send>
and i am running the script with below command :
C:\Program Files\Sipp_3.1>sipp -rsa 10.50.26.207:5060 -sf register_invite.xml 10
50.26.207:49152 -p 5062 -m 1
Regards,
Manaswini
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