Robert,
1. adding </recv> after my recv tags which have already been closed by
the / right before the closing > was not the right thing to do. SIPp is
not really good in detecting wrong syntax in the scenario file, which
explains why only the initial INVITE and 100 are displayed on the
runtime screen now after your modification.
In fact,
<recv attribute1="value1" attribute2="value2"*/*>
is equivalent to
<recv attribute1="value1" attribute2="value2">
</recv>
but shorter. Of course, the longer variant is the only possible one for
tags which contain a body, like all cases of send or if the recv
contains some action, like ereg.
2. the actual mistake causing SIPp to complain about an undefined label
was a typo: a "-" (dash) instead of "_" (underscore) in the label name
(after the <recv response="407"/>)
3. the bad UDP checksum is likely to cause the voicemail server not to
answer, but I'm not sure it is a SIPp issue - normally, an application
sends only the payload to a protocol socket, and the calculation of
checksums is a job of the protocol stack in the kernel. If your network
card eventually supports UDP checksum offloading, it may be that the
error doesn't actually exist, although in such case I would expect the
wrong value shown by tcpdump to be 0. To check that, you'll need to
capture the packets anywhere else at their way from the sender to the
recipient but at the sending machine (because there, the capturing point
is before the UDP checksum is calculated if the offloading is used). If
you can see the UDP checksum error after the packet has left the sending
machine, it is really there and likely to cause the receiving machine's
IP stack to drop the packet rather than deliver it to the application.
4. can you please run the tcpdump with -n option added, or place the
output ofgrep localhost /etc/hosts here?
It seems strange to me that the packet would have 127.0.0.1 as source
address as the localhost.localdomain fqdn suggests, but if it does, the
voicemail might to send the responses to itself if it looks at the
source address of the request packet rather than at the address given in
the Via header.
P.
Dne 4.8.2016 v 20:34 Remsik,Robert napsal(a):
Thank you for the help! This is indeed my base case copy+pasted, just
calling into voicemail and then either voicemail hangs up or I hang up.
I added the modifications you suggested but I had to add </recv> after
each recv tag to have the program not throw an error of: (The label
'auth_challenge_received' was not defined (index 2, next attribute)).
However now the scenario only lists an invite and 100 message, not
the full scenario that I was expecting.
Interestingly the INVITE lists a bad UDP checksum. That might help
explain why I'm not getting a response back from anything.
12:35:17.894298 IP (tos 0x0, ttl 64, id 62100, offset 0, flags [DF],
proto UDP (17), length 891)
localhost.localdomain.sip > 10.20.128.149.sip: [*bad udp cksum
0x128e -> 0xb0db!*] SIP, length: 863
INVITE sip:17120@10.20.128.149;user=phone;transport=tcp SIP/2.0
Via: SIP/2.0/UDP 129.82.3.26;branch=z9hG4bK-17371-1-0
From: "Robert Remsik" <sip:17...@otc.colostate.edu>;tag=1
To: <sip:17120@10.20.128.149;user=phone>
CSeq: 1 INVITE
Call-ID: 1-17371@129.82.3.26
Contact: <sip:17120@129.82.3.26;transport=UDP>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE,
SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomVVX-VVX_410-UA/4.1.8.0628
Accept-Language: en
Supported: 100rel,replaces
Allow-Events: conference,talk,hold
Max-Forwards: 70
Content-Type: application/sdp
Content-Length: 245
v=0
o=- 1470166919 1470166919 IN IP4 129.82.3.26
s=Polycom IP Phone
c=IN IP4 129.82.3.26
t=0 0
a=sendrecv
m=audio 6000 RTP/AVP 9 0 8 101
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
Messages Retrans Timeout
Unexpected-Msg
INVITE ----------> 1 0
100 <---------- 0 0 0 0
<recv response="100" optional="true"/>
</recv>
<recv response="401" optional="true" auth="true"
next="auth_challenge_received"/>
</recv>
<recv response="407" auth="true"/>
</recv>
<label id="auth_challenge-received"/>
</recv>
Robert Remsik
ACNS
Desk Phone: 970 491 7120
robert.rem...@colostate.edu
------------------------------------------------------------------------
*From:* sindelka <sinde...@ttc.cz>
*Sent:* Tuesday, August 2, 2016 9:12 PM
*To:* sipp-users@lists.sourceforge.net
*Subject:* Re: [Sipp-users] Troubleshooting a basic call
Hi Robert,
if the scenario you've provided is a verbatim copy of the one you
actually use, not an edited version for illustration purposes, the
trouble is that you blindly fire a series of requests without waiting
for any response from the server. Normally, the authentication challenge
comes in a 401 or 407 response to the first INVITE, and the generation
of the authentication response as requested by the [authentication]
keyword depends on the information from the authentication challenge.
So you must insert some optional and mandatory <recv> statements between
the first INVITE and first ACK - namely,
<recv response="100" optional="true"/>
<recv response="401" optional="true" auth="true"
next="auth_challenge_received"/>
<recv response="407" auth="true"/>
<label id="auth_challenge-received"/>
This ensures that regardless whether the sipx device sends the 100 or
not, and regardless whether it uses 401 or 407 to send the
authentication challenge, you'll handle it properly.
Also, remove the auth="true" AVPs from all the <send> blocks, they are
useless there.
On top of that, please <recv> optionally a 100, a 180, a 183 and then
compulsorily a 200 between sending the second INVITE (with the
authentication response) and sending the second ACK, otherwise you may
run into some other issues.
And the last point, if you specify user and password inside the
scenario, SIPp does not use the -au and -ap specified on the command line.
Pavel
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