The test case was as follow:
------------------------------------
(1) I called from a cell phone to sipX-based soft-phone.
(2) Upon receiving the call, the soft-phone played a prerecorded voice
message.
(3) After listening through a half of the message, I hung up the cell-phone
(4) The soft-phone was programmed in such way that it sends BYE message in
20 seconds after the voice message is sent
Expected results:
----------------------
The soft-phone has to receive BYE message from the cell phone after I hang
up the cell phone.
Actual results:
------------------
The soft-phone didn't receive BYE message and sent own BYE after 20 seconds
timeout.
Intentionally, I didn't capture RTP traffic, because I was able to hear the
prerecorded message sent by the soft-phone.
1 0.000000 10.1.1.9 64.24.35.208 SIP
Request: REGISTER sip:chi2-sr1-ha.starnetusa.net
2 0.506085 10.1.1.9 64.24.35.208 SIP
Request: REGISTER sip:chi2-sr1-ha.starnetusa.net
3 0.542242 64.24.35.208 10.1.1.9 SIP
Status: 401 Unauthorized (0 bindings)
4 0.550633 10.1.1.9 64.24.35.208 SIP
Request: REGISTER sip:chi2-sr1-ha.starnetusa.net
5 0.846945 64.24.35.208 10.1.1.9 SIP
Status: 401 Unauthorized (0 bindings)
6 0.897167 64.24.35.208 10.1.1.9 SIP
Status: 200 OK (1 bindings)
7 5.393642 64.24.35.208 10.1.1.9 SIP/SDP
Request: INVITE
sip:[EMAIL PROTECTED]:50170;LINEID=31740ec2dc3c1210d95634aaf29f63a4,
with session description
Internet Protocol, Src: 64.24.35.208 (64.24.35.208), Dst: 10.1.1.9 (10.1.1.9
)
User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 50170 (50170)
Session Initiation Protocol
Request-Line: INVITE
sip:[EMAIL PROTECTED]:50170;LINEID=31740ec2dc3c1210d95634aaf29f63a4
SIP/2.0
Message Header
Via: SIP/2.0/UDP 64.24.35.208;branch=0
Via: SIP/2.0/UDP 64.24.35.73:5060;branch=z9hG4bK3c10032d;rport=5060
From: "8472222222" <sip:[EMAIL PROTECTED]>;tag=as1a802ca9
To: <sip:[EMAIL PROTECTED]>
Contact: <sip:[EMAIL PROTECTED]>
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
User-Agent: US LEC VOIP
Max-Forwards: 16
Date: Mon, 26 Mar 2007 05:16:51 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 259
Message body
Session Description Protocol
8 5.396549 10.1.1.9 64.24.35.208 SIP
Status: 100 Trying
9 6.107420 10.1.1.9 64.24.35.208 SIP
Status: 180 Ringing
10 6.636539 10.1.1.9 64.24.35.208 SIP/SDP
Status: 200 OK, with session description
11 7.646438 10.1.1.9 64.24.35.208 SIP/SDP
Status: 200 OK, with session description
12 9.656544 10.1.1.9 64.24.35.208 SIP/SDP
Status: 200 OK, with session description
13 13.676523 10.1.1.9 64.24.35.208 SIP/SDP
Status: 200 OK, with session description
14 21.716276 10.1.1.9 64.24.35.208 SIP/SDP
Status: 200 OK, with session description
15 28.321108 10.1.1.9 64.24.35.208 SIP
Request: REGISTER sip:chi2-sr1-ha.starnetusa.net
16 29.168942 64.24.35.208 10.1.1.9 SIP
Status: 401 Unauthorized (0 bindings)
17 29.177208 10.1.1.9 64.24.35.208 SIP
Request: REGISTER sip:chi2-sr1-ha.starnetusa.net
18 29.430630 64.24.35.208 10.1.1.9 SIP
Status: 200 OK (1 bindings)
19 48.656751 10.1.1.9 64.24.35.73 SIP
Request: BYE sip:[EMAIL PROTECTED]
20 48.755317 10.1.1.9 64.24.35.73 SIP
Request: BYE sip:[EMAIL PROTECTED]
21 48.917136 64.24.35.73 10.1.1.9 SIP
Status: 200 OK
22 49.000808 64.24.35.73 10.1.1.9 SIP
Status: 200 OK
On 3/30/07, Alexander Chemeris <[EMAIL PROTECTED]> wrote:
Hello,
On 3/31/07, Aleksey Beregov <[EMAIL PROTECTED]> wrote:
> I was using Ethereal to capture communication between the cell and soft
> phones. Also our soft-phone logs all the events sent by API. Both
sources
> indicated that there was no BYE message.
> Thank you for advice to use WireShark. I will use it to debug this
issue
> further.
Wireshark is just a new name of Ethereal.
See http://wireshark.org/
--
Regards,
Alexander Chemeris.
SIPez LLC.
SIP VoIP, IM and Presence Consulting
http://www.SIPez.com
tel: +1 (617) 273-4000
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