I had tried that as well. It seems to kill SIPp with these errors dispalyed:
2 [main] sipp 9852 _cygtls::handle_exceptions: Exception:
STATUS_ACCESS_VIOLATION
1815 [main] sipp 9852 open_stackdumpfile: Dumping stack trace to
sipp.exe.stac kdump
1215547 [main] sipp 9852 _cygtls::handle_exceptions: Exception:
STATUS_ACCESS_VIOLATION
1230519 [main] sipp 9852 _cygtls::handle_exceptions: Error while dumping
state (probably corrupted stack)
On Mon, Jul 12, 2010 at 4:47 PM, Kalpan Doshi <kdo...@aumtech.com> wrote:
> Oh I see whats happening. Everything after the Authorization header is
> being interpreted as the content (SDP) including the remaining headers. If
> you look at the trace in a linux editor (or vim on windows), you can see
> that there are two '^M' characters at the end of the string indicating a
> double CRLF. In SIP, that means that is where the SDP begins. I am not sure
> why the Authorization header is putting a double CRLF.
>
> Here is what you can try:
>
> INVITE sip:1...@[remote_ip]:[remote_port] SIP/2.0
> Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
> From: <sip:1...@[local_ip]:[local_port]>;tag=[call_number]
> To: <sip:6165551...@[remote_ip]:[remote_port]>
> Call-ID: [call_id]
> CSeq: [cseq] INVITE
> Contact: sip:1...@[local_ip]:[local_port]
> Max-Forwards: 70
> Subject: Performance Test
> Content-Type: application/sdp
> Content-Length: [len]
> [field1] (Since this is adding a double CRLF, don't add an extra line
> after this before the SDP starts).
> 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
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
> a=ptime:20
>
> ]]>
> </send>
>
>
> Tim King wrote:
>
> Thanks again for the help.. Here is the updated scenario with the line
> feeds. If I take out the [field1] in the INVITE I get the loop of 407.. I
> have attached the trace files of the two calls.
>
>
> *Here is the updated scenario:*
> <?xml version="1.0" encoding="us-ascii"?>
> <scenario name="New_Call">
> <send retrans="500">
> <![CDATA[
> REGISTER sip:1...@[remote_ip]:[remote_port] SIP/2.0
> Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
> From: <sip:1...@[local_ip]:[local_port]>;tag=[call_number]
> To: <sip:1...@[remote_ip]:[remote_port]>
> Call-ID: [call_id]
> CSeq: [cseq] REGISTER
> Contact: sip:1...@[local_ip]:[local_port]
> Max-Forwards: 70
> Subject: Performance Test
> Content-Type: application/sdp
> Content-Length: [len]
> ]]>
> </send>
> <recv response="100" optional="true" />
> <recv response="401" auth="true" next="2" />
> <label id="2" />
> <send retrans="500">
> <![CDATA[
> REGISTER sip:1...@[remote_ip]:[remote_port] SIP/2.0
> Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
> From: <sip:1...@[local_ip]:[local_port]>;tag=[call_number]
> To: <sip:1...@[remote_ip]:[remote_port]>
> Call-ID: [call_id]
> CSeq: [cseq+1] REGISTER
> Contact: sip:1...@[local_ip]:[local_port];transport=UDP
> [field1]
> Max-Forwards: 70
> Subject: Performance Test
> Content-Type: application/sdp
> Content-Length: [len]
> ]]>
> </send>
> <recv response="200" crlf="true" />
> <send retrans="500">
> <![CDATA[
> INVITE sip:1...@[remote_ip]:[remote_port] SIP/2.0
> Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
> From: <sip:1...@[local_ip]:[local_port]>;tag=[call_number]
> To: <sip:6165551...@[remote_ip]:[remote_port]>
> Call-ID: [call_id]
> CSeq: [cseq] INVITE
> Contact: sip:1...@[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
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
> a=ptime:20
>
> ]]>
> </send>
> <recv response="100" optional="false" />
> <recv response="407" optional="false" next="3" />
> <label id="3" />
> <send>
> <![CDATA[
> ACK sip:6165551...@[remote_ip]:[remote_port] SIP/2.0
> Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
> From: <sip:1...@[local_ip]:[local_port]>;tag=[call_number]
> To: <sip:6165551...@[remote_ip]:[remote_port]>[peer_tag_param]
> Call-ID: [call_id]
> CSeq: [cseq+1] ACK
> Contact: sip:1...@[local_ip]:[local_port]
> Max-Forwards: 70
> Subject: Performance Test
> Content-Length: 0
> ]]>
> </send>
> <send retrans="500">
> <![CDATA[
> INVITE sip:1...@[remote_ip]:[remote_port] SIP/2.0
> Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
> From: <sip:1...@[local_ip]:[local_port]>;tag=[call_number]
> To: <sip:6165551...@[remote_ip]:[remote_port]>
> Call-ID: [call_id]
> CSeq: [cseq] INVITE
> Contact: sip:1...@[local_ip]:[local_port]
> [field1] <<-----------------------------------------------Taking that
> out does nto give the 415 but the instead I just loop with the 407 and
> invite.
> 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
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
> a=ptime:20
>
> ]]>
> </send>
> <recv response="100" optional="false" />
> <recv response="180" optional="false" />
> <recv response="200" crlf="false" />
> <send>
> <![CDATA[
> ACK sip:1...@[remote_ip]:[remote_port] SIP/2.0
> Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
> From: <sip:1...@[local_ip]:[local_port]>;tag=[call_number]
> To: <sip:6165551...@[remote_ip]:[remote_port]>[peer_tag_param]
> Call-ID: [call_id]
> CSeq: [cseq+1] ACK
> Contact: sip:s...@[local_ip]:[local_port]
> Max-Forwards: 70
> Subject: Performance Test
> Content-Length: 0
> ]]>
> </send>
> <pause milliseconds="50000" />
> <send retrans="500">
> <![CDATA[
> BYE sip:6165551...@[remote_ip]:[remote_port] SIP/2.0
> Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
> From: <sip:1...@[local_ip]:[local_port]>;tag=[call_number]
> To: <sip:6165551...@[remote_ip]:[remote_port]>[peer_tag_param]
> Call-ID: [call_id]
> CSeq: 2 BYE
> Contact: sip:s...@[local_ip]:[local_port]
> Max-Forwards: 70
> Subject: Performance Test
> Content-Length: 0
>
> ]]>
> </send>
> <label id="1" />
> <ResponseTimeRepartition value="10, 20, 30, 40, 50, 100, 150, 200" />
> <CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000" />
> </scenario>
>
>
> On Mon, Jul 12, 2010 at 4:28 PM, Kalpan Doshi <kdo...@aumtech.com> wrote:
>
>> Nevermind, I see that you have the full 415 message in the e-mail. It
>> doesn't give much additional information. The case may be that it is failing
>> with 415 even before it tries to do the authentication. Add a CRLF in your
>> first INVITE after ptime:20. See if that gets you the 100 message and then
>> the 407.
>>
>> Kalpan Doshi wrote:
>>
>> The '415 Unsupported Media Type' is usually caused by the Media Parameters
>> in the SDP. My theory is that the authentication is happening successfully,
>> the next step is to negotiate the media and that is where it is failing. You
>> can put the [field1] in the same place as the REGISTER (after Content-Type,
>> before Content-Length). Also, are you collecting the messages using the
>> -trace_msg? The full 415 message may give you additional information on why
>> its failing with Unsupported Media Type.
>>
>>
>> Tim King wrote:
>>
>> I was trying to figure out where to insert [field1] into the INVITE
>> message. I only get the "2010-07-12 16:06:01:609 1278965161.609105:
>> Aborting call on unexpected message for Call-Id '1-8...@192.168.0.10':
>> while expecting '100' (index 10), received 'SIP/2.0 415 Unsupported Media
>> Type" when I insert [field1] into the INVITE packet.
>>
>> On Mon, Jul 12, 2010 at 4:04 PM, Kalpan Doshi <kdo...@aumtech.com> wrote:
>>
>>> Tim,
>>>
>>> The Unsupported Media Type may be caused due to the fact that it cannot
>>> read the rtpmap attribute for PCMU. Try adding a CRLF after the last line of
>>> your SDP in the second INVITE. Your first INVITE and second INVITE have
>>> differences in the SDP (first one includes support for RFC 2833 payload
>>> 101). You should leave that in the second INVITE as well.
>>>
>>> Your second INVITE should look as follows:
>>>
>>>
>>> <send retrans="500">
>>> <![CDATA[
>>> INVITE sip:1...@[remote_ip]:[remote_port] SIP/2.0
>>> Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
>>> From: <sip:1...@[local_ip]:[local_port]>;tag=[call_number]
>>> To: <sip:6165551...@[remote_ip]:[remote_port]>
>>> Call-ID: [call_id]
>>> CSeq: [cseq] INVITE
>>> Contact: sip:1...@[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
>>> a=rtpmap:101 telephone-event/8000
>>> a=fmtp:101 0-15
>>> a=ptime:20
>>>
>>> ]]>
>>> </send>
>>>
>>>
>>> Regards,
>>> Kalpan
>>>
>>> Tim King wrote:
>>>
>>> This is the error I am getting now.
>>>
>>> -------------------------+---------------------------+--------------------------
>>> Call Length | 00:00:00:000 | 00:00:00:016
>>> ------------------------------ Test Terminated
>>> --------------------------------
>>>
>>> 2010-07-12 14:52:15:256 1278960735.256563: Aborting call on
>>> unexpected message for Call-Id '1-4...@192.168.0.10': while expecting
>>> '100' (index 6), received 'SIP/2.0 415 Unsupported Media Type
>>>
>>> Via: SIP/2.0/UDP 192.168.0.10:5060;branch=z9hG4bK-4168-1-5
>>> From: <sip:1...@192.168.0.10:5060>;tag=1
>>> To: <sip:6165551...@192.168.0.79:5060>;tag=20ej9Fg3ScBgg
>>> Call-ID: 1-4...@192.168.0.10
>>> CSeq: 4 INVITE
>>> User-Agent: FreeSWITCH-mod_sofia/1.0.head-svn-17782
>>> Accept: application/sdp
>>> Accept-Encoding:
>>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
>>> REGISTER, REFER,
>>> NOTIFY, PUBLISH, SUBSCRIBE
>>> Supported: timer, precondition, path, replaces
>>> Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla,
>>> include-s
>>> ession-description, presence.winfo, message-summary, refer
>>> Content-Length: 0
>>>
>>> '.
>>>
>>> Here is the updated XML Scenario:
>>>
>>> <?xml version="1.0" encoding="us-ascii"?>
>>> <scenario name="New_Call">
>>> <send retrans="500">
>>> <![CDATA[
>>> REGISTER sip:1...@[remote_ip]:[remote_port] SIP/2.0
>>> Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
>>> From: <sip:1...@[local_ip]:[local_port]>;tag=[call_number]
>>> To: <sip:1...@[remote_ip]:[remote_port]>
>>> Call-ID: [call_id]
>>> CSeq: [cseq] REGISTER
>>> Contact: sip:1...@[local_ip]:[local_port]
>>> Max-Forwards: 70
>>> Subject: Performance Test
>>> Content-Type: application/sdp
>>> Content-Length: [len]
>>> ]]>
>>> </send>
>>> <recv response="100" optional="true" />
>>> <recv response="401" auth="true" next="2" />
>>> <label id="2" />
>>> <send retrans="500">
>>> <![CDATA[
>>> REGISTER sip:1...@[remote_ip]:[remote_port] SIP/2.0
>>> Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
>>> From: <sip:1...@[local_ip]:[local_port]>;tag=[call_number]
>>> To: <sip:1...@[remote_ip]:[remote_port]>
>>> Call-ID: [call_id]
>>> CSeq: [cseq+1] REGISTER
>>> Contact: sip:1...@[local_ip]:[local_port];transport=UDP
>>> Max-Forwards: 70
>>> Subject: Performance Test
>>> Content-Type: application/sdp
>>> [field1]
>>> Content-Length: [len]
>>> ]]>
>>> </send>
>>> <recv response="200" crlf="true" />
>>> <send retrans="500">
>>> <![CDATA[
>>> INVITE sip:6165551...@[remote_ip]:[remote_port] SIP/2.0
>>> Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
>>> From: <sip:1...@[local_ip]:[local_port]>;tag=[call_number]
>>> To: <sip:6165551...@[remote_ip]:[remote_port]>
>>> Call-ID: [call_id]
>>> CSeq: [cseq] INVITE
>>> Contact: sip:1...@[local_ip]:[local_port];transport=UDP
>>> [field1]
>>> Max-Forwards: 70
>>> Subject: Performance Test
>>> Content-Type: application/sdp
>>> Content-Length: [len]
>>>
>>> v=0
>>> o=111 843670094 843670094 IN IP4 [local_ip]
>>> s=-
>>> c=IN IP4 [local_ip]
>>> t=0 0
>>> a=sendrecv
>>> m=audio [media_port] RTP/AVP 0 101
>>> a=rtpmap:0 PCMU/8000
>>> a=rtpmap:101 telephone-event/8000
>>> a=fmtp:101 0-15
>>> a=ptime:20]]>
>>> </send>
>>> <recv response="100" optional="false" />
>>> <recv response="407" optional="false" next="3" />
>>> <label id="3" />
>>> <send>
>>> <![CDATA[
>>> ACK sip:6165551...@[remote_ip]:[remote_port] SIP/2.0
>>> Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
>>> From: <sip:1...@[local_ip]:[local_port]>;tag=[call_number]
>>> To: <sip:6165551...@[remote_ip]:[remote_port]>[peer_tag_param]
>>> Call-ID: [call_id]
>>> CSeq: [cseq+1] ACK
>>> Contact: sip:1...@[local_ip]:[local_port]
>>> Max-Forwards: 70
>>> Subject: Performance Test
>>> Content-Length: 0
>>> ]]>
>>> </send>
>>> <send retrans="500">
>>> <![CDATA[
>>> INVITE sip:1...@[remote_ip]:[remote_port] SIP/2.0
>>> Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
>>> From: <sip:1...@[local_ip]:[local_port]>;tag=[call_number]
>>> To: <sip:6165551...@[remote_ip]:[remote_port]>
>>> Call-ID: [call_id]
>>> CSeq: [cseq] INVITE
>>> Contact: sip:1...@[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="100" optional="false" />
>>> <recv response="180" optional="false" />
>>> <recv response="200" crlf="false" />
>>> <send>
>>> <![CDATA[
>>> ACK sip:1...@[remote_ip]:[remote_port] SIP/2.0
>>> Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
>>> From: <sip:1...@[local_ip]:[local_port]>;tag=[call_number]
>>> To: <sip:6165551...@[remote_ip]:[remote_port]>[peer_tag_param]
>>> Call-ID: [call_id]
>>> CSeq: [cseq+1] ACK
>>> Contact: sip:s...@[local_ip]:[local_port]
>>> Max-Forwards: 70
>>> Subject: Performance Test
>>> Content-Length: 0
>>> ]]>
>>> </send>
>>> <pause milliseconds="50000" />
>>> <send retrans="500">
>>> <![CDATA[
>>> BYE sip:6165551...@[remote_ip]:[remote_port] SIP/2.0
>>> Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
>>> From: <sip:1...@[local_ip]:[local_port]>;tag=[call_number]
>>> To: <sip:6165551...@[remote_ip]:[remote_port]>[peer_tag_param]
>>> Call-ID: [call_id]
>>> CSeq: 2 BYE
>>> Contact: sip:s...@[local_ip]:[local_port]
>>> Max-Forwards: 70
>>> Subject: Performance Test
>>> Content-Length: 0
>>>
>>> ]]>
>>> </send>
>>> <label id="1" />
>>> <ResponseTimeRepartition value="10, 20, 30, 40, 50, 100, 150, 200" />
>>> <CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000" />
>>> </scenario>
>>>
>>> ------------------------------
>>>
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>>
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