Posted: Mon Mar 17, 2003 11:40 pm Post subject: [Asterisk-Dev] Protos SIP Test
Hi, all
I have tested the newest CVS version * with PROTOS SIP Test-Suite (see
http://www.ee.oulu.fi/research/ouspg/protos/testing/c07/sip/ ,
http://www.cert.org/advisories/CA-2003-06.html )
It sends various INVITE test-cases (4526 testcases). The first results are:
Test 1: java -jar c07-sip-r1.jar -touri
user@asterisk_server -teardown -validcase
INVITE test-case
CANCEL
ACK for the teardown
valid INVITE
CANCEL for the valid INVITE
ACK for the valid INVITE teardown
Test is ok.
Test 2: : java -jar c07-sip-r1.jar -touri user@asterisk_server -fromuri
user@1.2.3.4
Only INVITE test-case
Everything is ok, but the number of active sip channels grows during this
test, and when it == 1003 asterisk stops answer :
WARNING[7176] : File rtp.c, Line 533 (ast_rtp_new): Unable to allocate
socket: Too many open files
WARNING[7176] : File chan_sip.c, Line 1167 (sip_alloc): Unable to create RTP
session: Too many open files
My sip context is very simple (only for test):
[incoming]
exten => 1001,1,Dial,SIP/BYEXTENSION,20
exten => t,1,Hangup
exten => i,1,Hangup
It seems, that * creates sip channels, and don't destroy it . Any ideas ?
Posted: Tue Mar 18, 2003 12:37 am Post subject: [Asterisk-Dev] Protos SIP Test
Does it not tear down the calls? Does "sip show channels" show the
channels are still there?
Mark
On Tue, 18 Mar 2003, Nickolay Shestakov wrote:
Quote:
Hi, all
I have tested the newest CVS version * with PROTOS SIP Test-Suite (see
http://www.ee.oulu.fi/research/ouspg/protos/testing/c07/sip/ ,
http://www.cert.org/advisories/CA-2003-06.html )
It sends various INVITE test-cases (4526 testcases). The first results are:
Test 1: java -jar c07-sip-r1.jar -touri
user@asterisk_server -teardown -validcase
INVITE test-case
CANCEL
ACK for the teardown
valid INVITE
CANCEL for the valid INVITE
ACK for the valid INVITE teardown
Test is ok.
Test 2: : java -jar c07-sip-r1.jar -touri user@asterisk_server -fromuri
user@1.2.3.4
Only INVITE test-case
Everything is ok, but the number of active sip channels grows during this
test, and when it == 1003 asterisk stops answer :
WARNING[7176] : File rtp.c, Line 533 (ast_rtp_new): Unable to allocate
socket: Too many open files
WARNING[7176] : File chan_sip.c, Line 1167 (sip_alloc): Unable to create RTP
session: Too many open files
My sip context is very simple (only for test):
[incoming]
exten => 1001,1,Dial,SIP/BYEXTENSION,20
exten => t,1,Hangup
exten => i,1,Hangup
It seems, that * creates sip channels, and don't destroy it . Any ideas ?
Posted: Tue Mar 18, 2003 1:18 am Post subject: [Asterisk-Dev] Protos SIP Test
Yes, "sip show channels" show the channels are active.
And, now I've noticed, that in my simple config: 10.0.0.1- Asterisk server,
10.0.0.2 -Cisco 7960, 10.0.0.3 -eStara Softphone
after the first call "sip show channels" shows the 1 active SIP channel,
Peer - 10.0.0.1, Username (None)...
after the second - 2 active SIP channels, Peer - 10.0.0.1, 10.0.01
and so on...
What's is the reason may be?
Nickolay
Test 1: java -jar c07-sip-r1.jar -touri
user@asterisk_server -teardown -validcase
INVITE test-case
CANCEL
ACK for the teardown
valid INVITE
CANCEL for the valid INVITE
ACK for the valid INVITE teardown
Test is ok.
Test 2: : java -jar c07-sip-r1.jar -touri user@asterisk_server -fromuri
user@1.2.3.4
Only INVITE test-case
Everything is ok, but the number of active sip channels grows during this
test, and when it == 1003 asterisk stops answer :
WARNING[7176] : File rtp.c, Line 533 (ast_rtp_new): Unable to allocate
socket: Too many open files
WARNING[7176] : File chan_sip.c, Line 1167 (sip_alloc): Unable to create
RTP
Quote:
session: Too many open files
My sip context is very simple (only for test):
[incoming]
exten => 1001,1,Dial,SIP/BYEXTENSION,20
exten => t,1,Hangup
exten => i,1,Hangup
It seems, that * creates sip channels, and don't destroy it . Any ideas ?
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