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Posted to dev@openmeetings.apache.org by Markus van Aalst <ma...@googlemail.com> on 2013/08/21 15:03:19 UTC
Fwd: SIP transport leaves room every second
Hi,
I have seen there has been a similar problem on july so I investigated some
time in my configs but still did not found a solution.
*Background:*
I have installed OM + Asterisk in a Amazon EC2 machine for some tests.
After some time I now got a connection to asterisk and OM. So I am able to
make SIP test calls with asterisk only and join conferences in OM via web
browser separately.
I have tried to connect asterisk and OM with these instructions:
http://openmeetings.apache.org/red5sip-integration_2.1.html
After first test calls with linphone I got several errors, so I have
checked my asterisk *.conf files and corrected some of the errors.
Afterwards I tried to restart the services in the following order:
1) red5
2) asterisk
3) red5sip
Now I have got the following phenomena. When I try to start red5sip service
(parallel I have a conf room running in the web browser with SIP
activated), red5sip service trys to connect to asterisk. While I can see
SIP Transport joining and leaving the conf room within a second, the
following logs were generated in background:
*asterisk CLI (asterisk -rvvvv):*
Using SIP VIDEO CoS mark 6
Using SIP RTP CoS mark 5
Executing [4009@rooms-red5sip:1] [1;36mGotoIf [0m("
[1;35mSIP/red5sip_user-0000002d [0m", " [1;35m0?ok:notavail [0m") in new
stack
Goto (rooms-red5sip,4009,3)
Executing [4009@rooms-red5sip:3] [1;36mHangup [0m("
[1;35mSIP/red5sip_user-0000002d [0m", " [1;35m [0m") in new stack
Spawn extension (rooms-red5sip, 4009, 3) exited non-zero on
'SIP/red5sip_user-0000002d'
[Aug 21 08:59:58] [1;31mWARNING [0m[10502]: [1;37mchan_sip.c [0m:
[1;37m4174 [0m [1;37mretrans_pkt [0m: Retransmission timeout reached on
transmission 277125004182@172.31.42.81 for seqno 1 (Critical Response) --
See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 31996ms with no response
Manager 'openmeetings' logged on from 127.0.0.1
Manager 'openmeetings' logged off from 127.0.0.1
Manager 'openmeetings' logged on from 127.0.0.1
Manager 'openmeetings' logged off from 127.0.0.1
Manager 'openmeetings' logged on from 127.0.0.1
Manager 'openmeetings' logged off from 127.0.0.1
Manager 'openmeetings' logged on from 127.0.0.1
....
*red5sip log:*
*
*
21 Aug 08:51:32 - [INFO ] o.r.c.n.r.BaseRTMPClientHandler: rtmp://
127.0.0.1:1935/openmeetings/9
21 Aug 08:51:32 - [INFO ] o.r.s.n.r.RTMPHandshake: Processing server
response for encryption
21 Aug 08:51:32 - [INFO ] o.r.s.n.r.RTMPHandshake: Type 0 digest comparison
success
21 Aug 08:51:32 - [INFO ] o.r.s.n.r.RTMPHandshake: server response part 2
validation success, is Flash Player v9 handshake
21 Aug 08:56:26 - [INFO ] o.r.c.n.r.BaseRTMPClientHandler: rtmp://
127.0.0.1:1935/openmeetings/0
21 Aug 08:56:28 - [INFO ] o.r.s.n.r.RTMPHandshake: Processing server
response for encryption
21 Aug 08:56:28 - [INFO ] o.r.s.n.r.RTMPHandshake: Type 0 digest comparison
success
21 Aug 08:56:28 - [INFO ] o.r.s.n.r.RTMPHandshake: server response part 2
validation success, is Flash Player v9 handshake
21 Aug 08:56:29 - [WARN ] n.s.e.c.ConfigurationFactory: No configuration
found. Configuring ehcache from ehcache-failsafe.xml found in the
classpath:
jar:file:/opt/red5sip/out/lib/ehcache-core-jar-2.5.0.jar!/ehcache-failsafe.xml
21 Aug 08:56:31 - [INFO ] o.r.c.n.r.BaseRTMPClientHandler: rtmp://
127.0.0.1:1935/openmeetings/9
21 Aug 08:56:31 - [INFO ] o.r.s.n.r.RTMPHandshake: Processing server
response for encryption
21 Aug 08:56:31 - [INFO ] o.r.s.n.r.RTMPHandshake: Type 0 digest comparison
success
21 Aug 08:56:31 - [INFO ] o.r.s.n.r.RTMPHandshake: server response part 2
validation success, is Flash Player v9 handshake
21 Aug 08:56:33 - [INFO ] n.s.e.u.UpdateChecker: New update(s) found: 2.6.5
[
http://www.terracotta.org/confluence/display/release/Release+Notes+Ehcache+Core+2.6].
Please check http://ehcache.org for the latest version.
21 Aug 08:56:37 - [INFO ] o.r.c.n.r.BaseRTMPClientHandler: rtmp://
127.0.0.1:1935/openmeetings/9
21 Aug 08:56:37 - [INFO ] o.r.s.n.r.RTMPHandshake: Processing server
response for encryption
21 Aug 08:56:37 - [INFO ] o.r.s.n.r.RTMPHandshake: Type 0 digest comparison
success
21 Aug 08:56:37 - [INFO ] o.r.s.n.r.RTMPHandshake: server response part 2
validation success, is Flash Player v9 handshake
21 Aug 08:56:40 - [INFO ] o.r.c.n.r.BaseRTMPClientHandler: rtmp://
127.0.0.1:1935/openmeetings/9
21 Aug 08:56:40 - [INFO ] o.r.s.n.r.RTMPHandshake: Processing server
response for encryption
21 Aug 08:56:40 - [INFO ] o.r.s.n.r.RTMPHandshake: Type 0 digest comparison
success
21 Aug 08:56:40 - [INFO ] o.r.s.n.r.RTMPHandshake: server response part 2
validation success, is Flash Player v9 handshake
21 Aug 08:56:42 - [WARN ] o.r.c.n.r.RTMPMinaIoHandler: Exception caught null
21 Aug 08:56:47 - [INFO ] o.r.c.n.r.BaseRTMPClientHandler: rtmp://
127.0.0.1:1935/openmeetings/9
21 Aug 08:56:47 - [INFO ] o.r.s.n.r.RTMPHandshake: Processing server
response for encryption
21 Aug 08:56:47 - [INFO ] o.r.s.n.r.RTMPHandshake: Type 0 digest comparison
success
*asterisk (/var/log/asterisk/message) log:*
*
*
[Aug 21 08:41:54] Asterisk 11.5.0 built by root @ ip-172X-42-X on a x86_64
running Linux on 2013-08-20 11:31:54 UTC
[Aug 21 08:41:54] NOTICE[10470] loader.c: 2 modules will be loaded.
[Aug 21 08:41:54] NOTICE[10470] res_odbc.c: Connecting asterisk
[Aug 21 08:41:54] NOTICE[10470] res_odbc.c: res_odbc: Connected to asterisk
[asterisk-connector]
[Aug 21 08:41:54] NOTICE[10470] res_odbc.c: Registered ODBC class
'asterisk' dsn->[asterisk-connector]
[Aug 21 08:41:54] NOTICE[10470] res_odbc.c: res_odbc loaded.
[Aug 21 08:41:54] NOTICE[10470] config.c: Registered Config Engine odbc
[Aug 21 08:41:55] NOTICE[10470] cdr.c: CDR simple logging enabled.
[Aug 21 08:41:55] NOTICE[10470] loader.c: 195 modules will be loaded.
[Aug 21 08:41:55] NOTICE[10470] res_smdi.c: No SMDI interfaces are
available to listen on, not starting SMDI listener.
[Aug 21 08:41:55] NOTICE[10470] config.c: Registered Config Engine sqlite3
[Aug 21 08:41:55] WARNING[10470] chan_sip.c: !!! PLEASE NOTE: Setting 'nat'
for a peer/user that differs from the global setting can make
[Aug 21 08:41:55] WARNING[10470] chan_sip.c: !!! the name of that peer/user
discoverable by an attacker. Replies for non-existent peers/users
[Aug 21 08:41:55] WARNING[10470] chan_sip.c: !!! will be sent to a
different port than replies for an existing peer/user. If at all possible,
[Aug 21 08:41:55] WARNING[10470] chan_sip.c: !!! use the global 'nat'
setting and do not set 'nat' per peer/user.
[Aug 21 08:41:55] WARNING[10470] chan_sip.c: !!! (config
category='red5sip_user' global force_rport='No' peer/user force_rport='Yes')
[Aug 21 08:41:55] WARNING[10470] sip/config_parser.c: nat=yes is
deprecated, use nat=force_rport,comedia instead
[Aug 21 08:41:55] WARNING[10470] chan_sip.c: !!! PLEASE NOTE: Setting 'nat'
for a peer/user that differs from the global setting can make
[Aug 21 08:41:55] WARNING[10470] chan_sip.c: !!! the name of that peer/user
discoverable by an attacker. Replies for non-existent peers/users
[Aug 21 08:41:55] WARNING[10470] chan_sip.c: !!! will be sent to a
different port than replies for an existing peer/user. If at all possible,
[Aug 21 08:41:55] WARNING[10470] chan_sip.c: !!! use the global 'nat'
setting and do not set 'nat' per peer/user.
[Aug 21 08:41:55] WARNING[10470] chan_sip.c: !!! (config category='2000'
global force_rport='No' peer/user force_rport='Yes')
[Aug 21 08:41:55] WARNING[10470] sip/config_parser.c: nat=yes is
deprecated, use nat=force_rport,comedia instead
[Aug 21 08:41:55] WARNING[10470] chan_sip.c: !!! PLEASE NOTE: Setting 'nat'
for a peer/user that differs from the global setting can make
[Aug 21 08:41:55] WARNING[10470] chan_sip.c: !!! the name of that peer/user
discoverable by an attacker. Replies for non-existent peers/users
[Aug 21 08:41:55] WARNING[10470] chan_sip.c: !!! will be sent to a
different port than replies for an existing peer/user. If at all possible,
[Aug 21 08:41:55] WARNING[10470] chan_sip.c: !!! use the global 'nat'
setting and do not set 'nat' per peer/user.
[Aug 21 08:41:55] WARNING[10470] chan_sip.c: !!! (config category='3000'
global force_rport='No' peer/user force_rport='Yes')
[Aug 21 08:41:55] NOTICE[10470] chan_skinny.c: Configuring skinny from
skinny.conf
[Aug 21 08:41:55] NOTICE[10470] confbridge/conf_config_parser.c: Adding
default_user profile to app_confbridge
[Aug 21 08:41:55] WARNING[10470] pbx.c: Extension '_400X!' priority 5 in
'rooms', label 'ok' already in use at priority 2
[Aug 21 08:41:55] NOTICE[10470] pbx_ael.c: Starting AEL load process.
[Aug 21 08:41:55] NOTICE[10470] pbx_ael.c: AEL load process: parsed config
file name '/etc/asterisk/extensions.ael'.
[Aug 21 08:41:55] NOTICE[10470] pbx_ael.c: AEL load process: checked config
file name '/etc/asterisk/extensions.ael'.
[Aug 21 08:41:55] NOTICE[10470] pbx_ael.c: AEL load process: compiled
config file name '/etc/asterisk/extensions.ael'.
[Aug 21 08:41:55] NOTICE[10470] pbx_ael.c: AEL load process: merged config
file name '/etc/asterisk/extensions.ael'.
[Aug 21 08:41:55] NOTICE[10470] pbx_ael.c: AEL load process: verified
config file name '/etc/asterisk/extensions.ael'.
[Aug 21 08:51:18] WARNING[10502] chan_sip.c: Retransmission timeout reached
on transmission 560674983284@172.X.42.X for seqno 2 (Critical Response) --
See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 31999ms with no response
[Aug 21 08:51:26] WARNING[10502] chan_sip.c: Retransmission timeout reached
on transmission 022417704653@172.X.42.X for seqno 1 (Critical Response) --
See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 32000ms with no response
[Aug 21 08:51:26] WARNING[10502] chan_sip.c: Retransmission timeout reached
on transmission 022417704653@172.X.42.X for seqno 1 (Critical Response) --
See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 32000ms with no response
[Aug 21 08:51:27] WARNING[10502] chan_sip.c: Retransmission timeout reached
on transmission 022417704653@172.X.42.X for seqno 1 (Critical Response) --
See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 32000ms with no response
[Aug 21 08:51:27] WARNING[10502] chan_sip.c: Retransmission timeout reached
on transmission 022417704653@172.X.42.X for seqno 1 (Critical Response) --
See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 32000ms with no response
[Aug 21 08:51:28] WARNING[10502] chan_sip.c: Retransmission timeout reached
on transmission 022417704653@172.X.42.X for seqno 1 (Critical Response) --
See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 31978ms with no response
Any ideas?
Regards Markus
markus.van.aalst@gmail.com