Ruhi ASLAN
Stagiaire ST40 - NOC/Operation
VTX SERVICES SA
Une société du groupe VTX Telecom
================================================================
Tél. direct : 021 721 12 18
Av. de Lavaux 101 - 1009 Pully
http://www.vtx.ch <http://www.vtx.ch/> - ruhi.as...@vtx-telecom.ch
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VTX, votre partenaire telecom proche de vous !
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________________________________
De : Ruhi Aslan
Envoyé : vendredi, 18. juin 2010 18:07
À : sipp-users-requ...@lists.sourceforge.net
Objet : RE: sipp uas with -m 1 don't see the invite
Hi, I forward to request adresse, not sure witch is the good one ...
Since yesterday I'm looking in the code, and I don't find any clue about the
exacte problem.. But it's look like scenario.cpp contains the bug with a lot
conditionnal branch on MODE_UAS, it 's easy to make a mistake...
If you have same trouble and you write/got the patch, I'm eager to know ... :¬)
Ruhi ASLAN
Stagiaire ST40 - NOC/Operation
VTX SERVICES SA
Une société du groupe VTX Telecom
================================================================
Tél. direct : 021 721 12 18
Av. de Lavaux 101 - 1009 Pully
http://www.vtx.ch <http://www.vtx.ch/> - ruhi.as...@vtx-telecom.ch
----------------------------------------------------------------
VTX, votre partenaire telecom proche de vous !
================================================================
________________________________
De : Ruhi Aslan
Envoyé : mercredi, 16. juin 2010 15:13
À : sipp-users@lists.sourceforge.net
Objet : sipp uas with -m 1 don't see the invite
Hi all sipp users,
During my test with this tool, I notice an issue with -m 1 option in UAS mode.
In fact, when I use -m option for example
sipp -sn uas -m 20
sipp -sn uac localhost
In that case I have not this problem. But When I register a sipp uas to my
asterisk :
<register process with proxy>
REGISTER sip:Proxy SIP/2.0
Via: SIP/2.0/UDP MyPC:5060;branch=z9hG4bK-243-1-7
From: <sip:4...@mypc>;tag=11ddf
To: <sip:4...@mypc>
Call-ID: 1-24...@mypc
CSeq: 4 REGISTER
Contact: sip:4...@mypc:5060
Authorization: Digest
username="43",realm="voip",uri="sip:Proxy:5060",nonce="2ac9b9",response="7a6f1ea292baf",algorithm=MD5
Max-Forwards: 5
Expires: 300
User-Agent: SIPp/Linux
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE,
INFO
Content-Length: 0
SIP/2.0 200 OK
Via: SIP/2.0/UDP MyPC:5060;branch=z9hG4bK-24613-1-7
Record-Route: <sip:Proxy;lr=on;ftag=11ddf>
From: <sip:4...@mypc>;tag=11ddf
To: <sip:4...@mypc>;tag=as7252aa64
Call-ID: 1-24...@mypc
CSeq: 4 REGISTER
User-Agent: voipua
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Expires: 300
Contact: <sip:4...@asterisk:5060>;expires=300
Date: Wed, 16 Jun 2010 12:25:38 GMT
Content-Length: 0
Now my sipp uas is ready and I call with another real phone. If you look at the
invite below, it is correctly send to my sipp but my sipp don't see it.
In that case I used -m 1 option to end uas when it has 1 call. But I have the
same problem with sipp as uas mode :
sipp -sn uas -aa -m 1
I've got this below with retransmitions from my asterisk because sipp don't see
the INVITE :
INVITE sip:4...@mypc:5060 SIP/2.0
Record-Route:
<sip:Proxy;lr=on;ftag=as566237c6;vsf=AAAAAAAAAAAAAAAAAAAARF5bXh9CaVQ--;did=875.2a837>
Via: SIP/2.0/UDP Proxy;branch=z9hG4bK200c.d.0
Via: SIP/2.0/UDP ASterisk:5060;received=ASterisk;branch=z9hG4bK192;rport=5060
From: "11" <sip:1...@voip.vtx.ch>;tag=as566237c6
To: <sip:4...@mypc:5060>
Contact: <sip:1...@asterisk>
Call-ID: 393a9fcf212a05c477858...@asterisk
<mailto:393a9fcf212a05c4744bd80c27858...@asterisk>
CSeq: 102 INVITE
User-Agent: voipua
Max-Forwards: 69
Date: Wed, 16 Jun 2010 12:25:46 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Content-Type: application/sdp
Content-Length: 289
P-hint: outbound
v=0
o=root 12670 12670 IN IP4 ASterisk
s=session
c=IN IP4 ASterisk
t=0 0
m=audio 16482 RTP/AVP 8 18 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
I have used with and without tcpdump on 5060 to check if the problem is not
with linux port handling or whatever...
Why did my sipp see this Invite with -m 1 or -m 2 options ?
Beside this sipp -sn uas -aa works great, all call are successfull... :(
So if you have any suggestion, I will be happy to read it.
Ruhi ASLAN
Stagiaire ST40 - NOC/Operation
VTX SERVICES SA
Une société du groupe VTX Telecom
================================================================
Tél. direct : 021 721 12 18
Av. de Lavaux 101 - 1009 Pully
http://www.vtx.ch <http://www.vtx.ch/> - ruhi.as...@vtx-telecom.ch
----------------------------------------------------------------
VTX, votre partenaire telecom proche de vous !
================================================================
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