Help to fix error Voice connection timeout when use bbb-android to connect to voice conference

trương thế linh

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Mar 28, 2013, 9:59:27 PM3/28/13

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Hi all,

I'm using bigbluebutton version 0.80. 

I face the problem Voice connection timeout when using bbb-app on mobile.

Bellow is my network infor:

eth0      Link encap:Ethernet  HWaddr 06:9d:73:ec:3c:73
          inet addr:10.66.143.134  Bcast:10.66.143.191  Mask:255.255.255.192
          inet6 addr: fe80::49d:73ff:feec:3c73/64 Scope:Link
          UP BROADCAST RUNNING MULTICAST  MTU:1500  Metric:1
          RX packets:232833 errors:0 dropped:0 overruns:0 frame:0
          TX packets:95877 errors:0 dropped:0 overruns:0 carrier:0
          collisions:0 txqueuelen:1000
          RX bytes:284136704 (284.1 MB)  TX bytes:6860386 (6.8 MB)

eth1      Link encap:Ethernet  HWaddr 06:f2:f4:c4:9c:85
          inet addr:119.81.3.158  Bcast:119.81.3.159  Mask:255.255.255.248
          inet6 addr: fe80::4f2:f4ff:fec4:9c85/64 Scope:Link
          UP BROADCAST RUNNING MULTICAST  MTU:1500  Metric:1
          RX packets:147518 errors:0 dropped:0 overruns:0 frame:0
          TX packets:93839 errors:0 dropped:0 overruns:0 carrier:0
          collisions:0 txqueuelen:1000
          RX bytes:184554398 (184.5 MB)  TX bytes:19425938 (19.4 MB)

lo        Link encap:Local Loopback
          inet addr:127.0.0.1  Mask:255.0.0.0
          inet6 addr: ::1/128 Scope:Host
          UP LOOPBACK RUNNING  MTU:16436  Metric:1
          RX packets:17010 errors:0 dropped:0 overruns:0 frame:0
          TX packets:17010 errors:0 dropped:0 overruns:0 carrier:0
          collisions:0 txqueuelen:0
          RX bytes:3443983 (3.4 MB)  TX bytes:3443983 (3.4 MB)

my server public ip is: 119.81.3.158.

I can access bigbluebutton from address: http://119.81.3.158 from web

I have set ip: 119.81.3.158 on eth1 for my bigbluebutton.
By run command: bbb-conf --setip 119.81.3.158.

I also use this script: https://raw.github.com/mconf/installation-scripts/master/bbb-deploy/enable-mobile-fs.sh

but it doesn't work.


Bellow are my result when run command: bbb-conf --check:

BigBlueButton Server 0.8-beta-4 (951)
                    Kernel version: 2.6.32-350-ec2
                      Distribution: Ubuntu 10.04.4 LTS (32-bit)
                            Memory: 1040 MB

/var/www/bigbluebutton/client/conf/config.xml (bbb-client)
  		Port test (tunnel): 119.81.3.158
                              Red5: 119.81.3.158

/etc/nginx/sites-available/bigbluebutton (nginx)
                       server name: 119.81.3.158
                              port: 80
                    bbb-client dir: /var/www/bigbluebutton

/var/lib/tomcat6/webapps/bigbluebutton/WEB-INF/classes/bigbluebutton.properties (bbb-web)
                      bbb-web host: 119.81.3.158

/var/lib/tomcat6/webapps/demo/bbb_api_conf.jsp (API demos)
                           api url: 119.81.3.158

/usr/share/red5/webapps/bigbluebutton/WEB-INF/red5-web.xml (red5)
                  voice conference: FreeSWITCH

/usr/local/bigbluebutton/core/scripts/slides.yml (record and playback)
                     playback host: 119.81.3.158


** Potential problems described below **
# IP does not match:
#                           IP from ifconfig: 10.66.143.134
#   /etc/nginx/sites-available/bigbluebutton: 119.81.3.158
# Warning: API URL IPs do not match host:
#
#                                IP from ifconfig: 10.66.143.134
#  /var/lib/tomcat6/webapps/demo/bbb_api_conf.jsp: 119.81.3.158

# Error: The voice application failed to register with the sip server.
#   Try running: 
#
#      sudo bbb-conf --clean
#

# Error: The setting of (119.81.3.158) for sip.server.host in
#
#    /usr/share/red5/webapps/sip/WEB-INF/bigbluebutton-sip.properties
#
# does not match the local IP address (10.66.143.134).

# Error: FreeSWITCH is listening on IP address 127.0.0.1 for SIP calls, but 
# The IP address (119.81.3.158) set for sip.server.host.
#
# If your audio is not working (users click the  headset icon 
# and don't appear in the Listeners window, ensure FreeSWITCH uses
# the local loopback 127.0.0.1 address.  See
#   http://code.google.com/p/bigbluebutton/wiki/FAQ#Users_do_not_appear_in_the_listeners_window

# Warning: The value (119.81.3.158) for playback_host in
#
#    /usr/local/bigbluebutton/core/scripts/slides.yml
#
# does not match the local IP address (10.66.143.134).

# Warning: You are running BigBlueButton on a server with less than 2G of memory.  Your
# performance may suffer.

# Warning: The API demos are installed and accessible from:
#
#    http://119.81.3.158/
#
# Use the API demos test your BigBlueButton setup. To remove
#
#    sudo apt-get purge bbb-demo

Please help me.

trương thế linh

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Mar 28, 2013, 10:08:06 PM3/28/13

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Below are my content for some impotant confiure relate to sip, freeswitch:

bigbluebutton-sip.properties

===============================================

# The address of your FreeSWITCH/asterisk server

sip.server.host=119.81.3.158

sip.server.port=5070

sip.server.username=bbbuser

sip.server.password=secret

# The start/stop RTP port the application is going to use

# for the media stream.

# NOTE: This will also be used as SIP users to REGISTER with

# Asterisk. Therefore, make sure you have this range of users

# in your bbb_sip.conf.

# See http://code.google.com/p/bigbluebutton/source/browse/#svn/trunk/bbb-voice-conference/config/asterisk

# create-sip-users.sh script to create the users.

startAudioPort=15000

stopAudioPort=16383

# An extension pattern, in case your asterisk extensions.conf

# uses a naming convetion for your meeting rooms

# e.g. conf-85115 instead of just 85115

callExtensionPattern={0}

# If you want mjsip stack (red5/log/*access*.log) to minimize the amount of logs it

# generates, set this to a lower value (e.g. 3).

sipStackDebugLevel=3

===============================================

conference.conf.xml

===============================================

</advertise>

<caller-controls>

</group>

</caller-controls>

<!--Can be | delim of waste|mute|deaf|dist-dtmf waste will always transmit data to each channel

even during silence. dist-dtmf propagates dtmfs to all other members, but channel controls

via dtmf will be disabled. -->

<!--

-->

</profile>

<!-- param name="moh-sound" value="$${hold_music}"/>

</profile>

</profile>

</profile>

</profile>

</profiles>

</configuration>

===============================================

external

===============================================

<X-PRE-PROCESS cmd="include" data="external/*.xml"/>

</gateways>

<!--

-->

</aliases>

</domains>

<!-- used to share presence info across sofia profiles

manage-presence needs to be set to passive on this profile

if you want it to behave as if it were the internal profile

for presence.

-->

<!--

DO NOT USE HOSTNAMES, ONLY IP ADDRESSES IN THESE SETTINGS!

-->

<!--

Set the RTP timeout for 3 hours to prevent FS from hanging up bbb clients

who are muted and not sending any audio.

-->

</settings>

</profile>

===============================================

vars.xml

===============================================

<!-- Preprocessor Variables

These are introduced when configuration strings must be consistent across modules.

NOTICE: YOU CAN NOT COMMENT OUT AN X-PRE-PROCESS line, Remove the line instead.

WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING

YOU SHOULD CHANGE THIS default_password value if you don't want to be subject to any

toll fraud in the future. It's your responsibility to secure your own system.

This default config is used to demonstrate the feature set of FreeSWITCH.

WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING

-->

<X-PRE-PROCESS cmd="set" data="default_password=1234"/>

<X-PRE-PROCESS cmd="set" data="sound_prefix=$${sounds_dir}/en/us/callie"/>

<!--

This setting is what sets the default domain FreeSWITCH will use if all else fails.

FreeSWICH will default to $${local_ip_v4} unless changed. Changing this setting does

affect the sip authentication. Please review conf/directory/default.xml for more

information on this topic.

-->

<!--

<X-PRE-PROCESS cmd="set" data="local_ip_v4=127.0.0.1"/>

-->

<X-PRE-PROCESS cmd="set" data="domain=$${local_ip_v4}"/>

<X-PRE-PROCESS cmd="set" data="domain_name=$${domain}"/>

<X-PRE-PROCESS cmd="set" data="hold_music=local_stream://moh"/>

<X-PRE-PROCESS cmd="set" data="use_profile=internal"/>

<!--

Enable ZRTP globally you can override this on a per channel basis

http://wiki.freeswitch.org/wiki/ZRTP (on how to enable zrtp)

-->

<X-PRE-PROCESS cmd="set" data="zrtp_secure_media=true"/>

<!--

Examples of codec options: (module must be compiled and loaded)

codecname[@8000h|16000h|32000h[@XXi]]

XX is the frame size must be multples allowed for the codec

FreeSWITCH can support 10-120ms on some codecs.

We do not support exceeding the MTU of the RTP packet.

iLBC@30i - iLBC using mode=30 which will win in all cases.

DVI4@8000h@20i - IMA ADPCM 8kHz using 20ms ptime. (multiples of 10)

DVI4@16000h@40i - IMA ADPCM 16kHz using 40ms ptime. (multiples of 10)

speex@8000h@20i - Speex 8kHz using 20ms ptime.

speex@16000h@20i - Speex 16kHz using 20ms ptime.

speex@32000h@20i - Speex 32kHz using 20ms ptime.

BV16 - BroadVoice 16kb/s narrowband, 8kHz

BV32 - BroadVoice 32kb/s wideband, 16kHz

G7221@16000h - G722.1 16kHz (aka Siren 7)

G7221@32000h - G722.1C 32kHz (aka Siren 14)

CELT@32000h - CELT 32kHz, only 10ms supported

CELT@48000h - CELT 48kHz, only 10ms supported

GSM@40i - GSM 8kHz using 40ms ptime. (GSM is done in multiples of 20, Default is 20ms)

G722 - G722 16kHz using default 20ms ptime. (multiples of 10)

PCMU - G711 8kHz ulaw using default 20ms ptime. (multiples of 10)

PCMA - G711 8kHz alaw using default 20ms ptime. (multiples of 10)

G726-16 - G726 16kbit adpcm using default 20ms ptime. (multiples of 10)

G726-24 - G726 24kbit adpcm using default 20ms ptime. (multiples of 10)

G726-32 - G726 32kbit adpcm using default 20ms ptime. (multiples of 10)

G726-40 - G726 40kbit adpcm using default 20ms ptime. (multiples of 10)

AAL2-G726-16 - Same as G726-16 but using AAL2 packing. (multiples of 10)

AAL2-G726-24 - Same as G726-24 but using AAL2 packing. (multiples of 10)

AAL2-G726-32 - Same as G726-32 but using AAL2 packing. (multiples of 10)

AAL2-G726-40 - Same as G726-40 but using AAL2 packing. (multiples of 10)

LPC - LPC10 using 90ms ptime (only supports 90ms at this time in FreeSWITCH)

L16 - L16 isn't recommended for VoIP but you can do it. L16 can exceed the MTU rather quickly.

These are the passthru audio codecs:

G729 - G729 in passthru mode. (mod_g729)

G723 - G723.1 in passthru mode. (mod_g723_1)

AMR - AMR in passthru mode. (mod_amr)

These are the passthru video codecs: (mod_h26x)

H261 - H.261 Video

H263 - H.263 Video

H263-1998 - H.263-1998 Video

H263-2000 - H.263-2000 Video

H264 - H.264 Video

RTP Dynamic Payload Numbers currently used in FreeSWITCH and what for.

96 - AMR

97 - iLBC (30)

98 - iLBC (20)

99 - Speex 8kHz, 16kHz, 32kHz

100 -

101 - telephone-event

102 -

103 -

104 -

105 -

106 - BV16

107 - G722.1 (16kHz)

108 -

109 -

110 -

111 -

112 -

113 -

114 - CELT 32kHz, 48kHz

115 - G722.1C (32kHz)

116 -

117 - SILK 8kHz

118 - SILK 12kHz

119 - SILK 16kHz

120 - SILK 24kHz

121 - AAL2-G726-40 && G726-40

122 - AAL2-G726-32 && G726-32

123 - AAL2-G726-24 && G726-24

124 - AAL2-G726-16 && G726-16

125 -

126 -

127 - BV32

-->

<X-PRE-PROCESS cmd="set" data="global_codec_prefs=speex@16000h@20i,speex@8000h@20i,G7221@32000h,G7221@16000h,G722,PCMU,PCMA,GSM" />

<X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=speex@16000h@20i,PCMU,PCMA,GSM" />

<!-- BigBlueButton: To use mlaw , change the above two lines with data set to these values.

data="global_codec_prefs=PCMU,G722,PCMA,GSM"

data="outbound_codec_prefs=PCMU,G722,PCMA,GSM"

-->

<!--

xmpp_client_profile and xmpp_server_profile

xmpp_client_profile can be any string.

xmpp_server_profile is appended to "dingaling_" to form the database name

containing the "subscriptions" table.

used by: dingaling.conf.xml enum.conf.xml

-->

<X-PRE-PROCESS cmd="set" data="xmpp_client_profile=xmppc"/>

<X-PRE-PROCESS cmd="set" data="xmpp_server_profile=xmpps"/>

<!--

THIS IS ONLY USED FOR DINGALING

bind_server_ip

Can be an ip address, a dns name, or "auto".

This determines an ip address available on this host to bind.

If you are separating RTP and SIP traffic, you will want to have

use different addresses where this variable appears.

Used by: dingaling.conf.xml

-->

<X-PRE-PROCESS cmd="set" data="bind_server_ip=auto"/>

<!-- NOTICE NOTICE NOTICE NOTICE NOTICE NOTICE NOTICE NOTICE NOTICE NOTICE NOTICE

If you're going to load test FreeSWITCH please input real IP addresses

for external_rtp_ip and external_sip_ip

-->

<!-- external_rtp_ip

Can be an one of:

ip address: "12.34.56.78"

a stun server lookup: "stun:stun.server.com"

a DNS name: "host:host.server.com"

where fs.mydomain.com is a DNS A record-useful when fs is on

a dynamic IP address, and uses a dynamic DNS updater.

If unspecified, the bind_server_ip value is used.

Used by: sofia.conf.xml dingaling.conf.xml

-->

<X-PRE-PROCESS cmd="set" data="external_rtp_ip=stun:stun.freeswitch.org"/>

<!-- external_sip_ip

Used as the public IP address for SDP.

Can be an one of:

ip address: "12.34.56.78"

a stun server lookup: "stun:stun.server.com"

a DNS name: "host:host.server.com"

where fs.mydomain.com is a DNS A record-useful when fs is on

a dynamic IP address, and uses a dynamic DNS updater.

If unspecified, the bind_server_ip value is used.

Used by: sofia.conf.xml dingaling.conf.xml

-->

<X-PRE-PROCESS cmd="set" data="external_sip_ip=stun:stun.freeswitch.org"/>

<!-- unroll-loops

Used to turn on sip loopback unrolling.

-->

<X-PRE-PROCESS cmd="set" data="unroll_loops=true"/>

<!-- outbound_caller_id and outbound_caller_name

The caller ID telephone number we should use when calling out.

Used by: conference.conf.xml and user directory for default

outbound callerid name and number.

-->

<X-PRE-PROCESS cmd="set" data="outbound_caller_name=FreeSWITCH"/>

<X-PRE-PROCESS cmd="set" data="outbound_caller_id=0000000000"/>

<X-PRE-PROCESS cmd="set" data="call_debug=false"/>

<X-PRE-PROCESS cmd="set" data="console_loglevel=info"/>

<X-PRE-PROCESS cmd="set" data="default_areacode=918"/>

<X-PRE-PROCESS cmd="set" data="default_country=US"/>

<X-PRE-PROCESS cmd="set" data="uk-ring=%(400,200,400,450);%(400,2200,400,450)"/>

<X-PRE-PROCESS cmd="set" data="us-ring=%(2000,4000,440.0,480.0)"/>

<X-PRE-PROCESS cmd="set" data="fr-ring=%(1500,3500,440.0,0.0)"/>

<X-PRE-PROCESS cmd="set" data="rs-ring=%(1000,4000,425.0,0.0)"/>

<X-PRE-PROCESS cmd="set" data="ru-ring=%(800,3200,425,0)"/>

<X-PRE-PROCESS cmd="set" data="bong-ring=v=-7;%(100,0,941.0,1477.0);v=-7;>=2;+=.1;%(1400,0,350,440)"/>

<X-PRE-PROCESS cmd="set" data="sit=%(274,0,913.8);%(274,0,1370.6);%(380,0,1776.7)"/>

<!--

Setting up your default sip provider is easy.

Below are some values that should work in most cases.

These are for conf/directory/default/example.com.xml

-->

<X-PRE-PROCESS cmd="set" data="default_provider=example.com"/>

<X-PRE-PROCESS cmd="set" data="default_provider_username=joeuser"/>

<X-PRE-PROCESS cmd="set" data="default_provider_password=password"/>

<X-PRE-PROCESS cmd="set" data="default_provider_from_domain=example.com"/>

<X-PRE-PROCESS cmd="set" data="default_provider_register=false"/>

<X-PRE-PROCESS cmd="set" data="default_provider_contact=5000"/>

<!--

SIP and TLS settings. http://wiki.freeswitch.org/wiki/Tls

-->

<X-PRE-PROCESS cmd="set" data="sip_tls_version=tlsv1"/>

<X-PRE-PROCESS cmd="set" data="internal_auth_calls=true"/>

<X-PRE-PROCESS cmd="set" data="internal_sip_port=5090"/>

<X-PRE-PROCESS cmd="set" data="internal_tls_port=5061"/>

<X-PRE-PROCESS cmd="set" data="internal_ssl_enable=false"/>

<X-PRE-PROCESS cmd="set" data="internal_ssl_dir=$${base_dir}/conf/ssl"/>

<X-PRE-PROCESS cmd="set" data="external_auth_calls=false"/>

<X-PRE-PROCESS cmd="set" data="external_sip_port=5060"/>

<X-PRE-PROCESS cmd="set" data="external_tls_port=5081"/>

<X-PRE-PROCESS cmd="set" data="external_ssl_enable=false"/>

<X-PRE-PROCESS cmd="set" data="external_ssl_dir=$${base_dir}/conf/ssl"/>

</include>

===============================================

Felipe Cecagno

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Mar 30, 2013, 11:12:51 PM3/30/13

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Hi,

From your information:

# Error: The voice application failed to register with the sip server.
#   Try running: 
#
#      sudo bbb-conf --clean
#

If the voice application cannot register on the sip server, you shouldn't be able to join the voice conference on the web client. Can you confirm that the audio is working in your setup, before trying to use your Android device?

You could try to follow the hint and run "sudo bbb-conf --clean" and test it again. If the problem persists, please write back.

Regards,

--

Felipe Cecagno

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Chad Pilkey

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Mar 31, 2013, 11:46:48 AM3/31/13

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I would echo what Felipe said, you should be verifying in the Flash client that the audio still works after your changes.

Also, I don't think you have your settings correct in vars.xml. The part that reads:

<!--

<X-PRE-PROCESS cmd="set" data="local_ip_v4=127.0.0.1"/>

-->
You should uncomment the line and change the 127.0.0.1 to your public IP if you want to allow external calls.

Chad

Thế Linh Trương

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Apr 1, 2013, 12:10:56 AM4/1/13

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Hi all,

Thanks for your help.

I have made some change on var.xml set:

<X-PRE-PROCESS cmd="set" data="domain=119.81.3.158"/>

And change in /usr/share/red5/webapps/sip/WEB-INF/bigbluebutton-sip.properties set:

sip.server.host=127.0.0.1

I have test on flash client. And saw that voice conference was worked.

But when i access from bbb android. I still face error: voice connection time out.

Please help me. I want connect from bbb android to bigbluebutton.

2013/3/31 Chad Pilkey <capi...@gmail.com>

Chad Pilkey

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Apr 1, 2013, 12:15:47 PM4/1/13

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I think sip.server.host should be set to your public ip as well.

Thế Linh Trương

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Apr 1, 2013, 12:33:05 PM4/1/13

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When i change sip.server.host should be set to your public ip i can't even join voice conference from flash client.

Bellows are My network infor:

eth0      Link encap:Ethernet HWaddr 06:9d:73:ec:3c:73
          inet addr:10.66.143.134 Bcast:10.66.143.191 Mask:255.255.255.192
          inet6 addr: fe80::49d:73ff:feec:3c73/64 Scope:Link
          UP BROADCAST RUNNING MULTICAST MTU:1500 Metric:1

          RX packets:2470 errors:0 dropped:0 overruns:0 frame:0
          TX packets:2646 errors:0 dropped:0 overruns:0 carrier:0
          collisions:0 txqueuelen:1000
          RX bytes:248017 (248.0 KB) TX bytes:1179517 (1.1 MB)

eth1      Link encap:Ethernet HWaddr 06:f2:f4:c4:9c:85
          inet addr:119.81.3.158 Bcast:119.81.3.159 Mask:255.255.255.248
          inet6 addr: fe80::4f2:f4ff:fec4:9c85/64 Scope:Link
          UP BROADCAST RUNNING MULTICAST MTU:1500 Metric:1

          RX packets:1131969 errors:0 dropped:0 overruns:0 frame:0
          TX packets:955657 errors:0 dropped:0 overruns:0 carrier:0
          collisions:0 txqueuelen:1000
          RX bytes:83026665 (83.0 MB) TX bytes:109616923 (109.6 MB)

lo        Link encap:Local Loopback
          inet addr:127.0.0.1 Mask:255.0.0.0
          inet6 addr: ::1/128 Scope:Host
          UP LOOPBACK RUNNING MTU:16436 Metric:1

          RX packets:1905812 errors:0 dropped:0 overruns:0 frame:0
          TX packets:1905812 errors:0 dropped:0 overruns:0 carrier:0
          collisions:0 txqueuelen:0
          RX bytes:144167923 (144.1 MB) TX bytes:144167923 (144.1 MB)

When i run bbb-conf --check, i get bellow result:

BigBlueButton Server 0.8-beta-4 (951)
Kernel version: 2.6.32-351-ec2

My public ip is: 119.81.3.158

Please help me.

2013/4/1 Chad Pilkey <capi...@gmail.com>

Chad Pilkey

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Apr 1, 2013, 12:44:38 PM4/1/13

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I'm not sure what else to try, sorry. Hopefully someone else has more insight.

Felipe Cecagno

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Apr 1, 2013, 1:53:30 PM4/1/13

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See: https://code.google.com/p/bigbluebutton/issues/detail?id=1133#c8

There are only three things you need to change:

- Set sip.server.host on /usr/share/red5/webapp/sip/WEB-INF/bigbluebutton-sip.properties to your external IP

- in /opt/freeswitch/conf/vars.xml

<X-PRE-PROCESS cmd="set" data="domain=${local_ip_v4}"/>
(there is no definition for local_ip_v4)

- in /opt/freeswitch/conf/sip_profiles/external.xml

Then run "sudo bbb-conf --clean" and make sure that the audio is working fine on the web client. After that you should be able to use BBB-Android, just make sure that you have setup your firewall correctly - you must open the following ports:

- UDP 5060 (SIP) and UDP 16384-32768 (RTP)

This is the configuration needed to enable external SIP phones to call your BigBlueButton server, and it will enable BBB-Android connect to it as well.

--

Felipe Cecagno

Thế Linh Trương

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Apr 1, 2013, 11:04:02 PM4/1/13

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Hi all,

I have tried but it doesn't work.

Bellow is my network confiure:

eth0 Link encap:Ethernet HWaddr 06:9d:73:ec:3c:73

inet addr:10.66.143.134 Bcast:10.66.143.191 Mask:255.255.255.192

inet6 addr: fe80::49d:73ff:feec:3c73/64 Scope:Link

UP BROADCAST RUNNING MULTICAST MTU:1500 Metric:1

RX packets:4386 errors:0 dropped:0 overruns:0 frame:0

TX packets:4671 errors:0 dropped:0 overruns:0 carrier:0

collisions:0 txqueuelen:1000

RX bytes:439341 (439.3 KB) TX bytes:2091500 (2.0 MB)

eth1 Link encap:Ethernet HWaddr 06:f2:f4:c4:9c:85

inet addr:119.81.3.158 Bcast:119.81.3.159 Mask:255.255.255.248

inet6 addr: fe80::4f2:f4ff:fec4:9c85/64 Scope:Link

UP BROADCAST RUNNING MULTICAST MTU:1500 Metric:1

RX packets:1163087 errors:0 dropped:0 overruns:0 frame:0

TX packets:992087 errors:0 dropped:0 overruns:0 carrier:0

collisions:0 txqueuelen:1000

RX bytes:86190197 (86.1 MB) TX bytes:117733949 (117.7 MB)

---> This is my public ip. I only can access my server from external via this address: 119.81.3.159.

lo Link encap:Local Loopback

inet addr:127.0.0.1 Mask:255.0.0.0

inet6 addr: ::1/128 Scope:Host

UP LOOPBACK RUNNING MTU:16436 Metric:1

RX packets:1940106 errors:0 dropped:0 overruns:0 frame:0

TX packets:1940106 errors:0 dropped:0 overruns:0 carrier:0

collisions:0 txqueuelen:0

RX bytes:149048735 (149.0 MB) TX bytes:149048735 (149.0 MB)

As all of your guide i have change some confiure file as below:

/usr/share/red5/webapp/sip/WEB-INF/bigbluebutton-sip.properties:

====================================================

# The address of your FreeSWITCH/asterisk server sip.server.host=119.81.3.158 ---> I have change it to external ip. ( my external ip is 119.81.3.158 ) #sip.server.host=127.0.0.1 sip.server.port=5070 sip.server.username=bbbuser sip.server.password=secret # The start/stop RTP port the application is going to use # for the media stream. # NOTE: This will also be used as SIP users to REGISTER with # Asterisk. Therefore, make sure you have this range of users # in your bbb_sip.conf. # See http://code.google.com/p/bigbluebutton/source/browse/#svn/trunk/bbb-voice-conference/config/asterisk # create-sip-users.sh script to create the users. startAudioPort=15000 stopAudioPort=16383 # An extension pattern, in case your asterisk extensions.conf # uses a naming convetion for your meeting rooms # e.g. conf-85115 instead of just 85115 callExtensionPattern={0} # If you want mjsip stack (red5/log/*access*.log) to minimize the amount of logs it # generates, set this to a lower value (e.g. 3). sipStackDebugLevel=3

====================================================

/opt/freeswitch/conf/vars.xml:

====================================================

====================================================

I run command: bbb-conf --restart. But I still face error:

# sudo bbb-conf --clean # # Error: The setting of (119.81.3.158) for sip.server.host in # # /usr/share/red5/webapps/sip/WEB-INF/bigbluebutton-sip.properties # # does not match the local IP address (10.66.143.134). # Error: FreeSWITCH is listening on IP address 127.0.0.1 for SIP calls, but # The IP address (119.81.3.158) set for sip.server.host. # # If your audio is not working (users click the headset icon # and don't appear in the Listeners window, ensure FreeSWITCH uses # the local loopback 127.0.0.1 address. See # http://code.google.com/p/bigbluebutton/wiki/FAQ#Users_do_not_appear_in_the_listeners_window # Warning: The value (119.81.3.158) for playback_host in # # /usr/local/bigbluebutton/core/scripts/slides.yml # # does not match the local IP address (10.66.143.134). # Warning: You are running BigBlueButton on a server with less than 2G of memory. Your # performance may suffer. # Warning: The API demos are installed and accessible from: # # http://119.81.3.158/ # # Use the API demos test your BigBlueButton setup. To remove # # sudo apt-get purge bbb-demo

I have followed all your help. But it doesn't work. the bbb client don;t show list listener.

If any one have free time. Please help me login to my server. And make some change for me.

Thanks.

2013/4/2 Felipe Cecagno <fcec...@gmail.com>

Felipe Cecagno

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Apr 2, 2013, 9:03:45 AM4/2/13

to bigblueb...@googlegroups.com

I don't know why, but when I copied the instructions and pasted here, it replaced two $ by only one $. The proper definition is double $ in front of local_ip_v4. Please see this link for instructions: https://code.google.com/p/bigbluebutton/issues/detail?id=1133#c8

If you do exactly that and it doesn't work, I have no idea what's going on. I used this configuration many times and never had problems with it.

From your bbb-conf --clean output:

# Error: FreeSWITCH is listening on IP address 127.0.0.1 for SIP calls, but # The IP address (119.81.3.158) set for sip.server.host.

What you want is that FreeSWITCH listens to your external IP and not localhost. If after your changes it still says that FreeSWITCH is listening to localhost, try to define this on vars.xml:

<X-PRE-PROCESS cmd="set" data="local_ip_v4=119.81.3.158"/>

And then run bbb-conf --clean again.

--

Felipe Cecagno

abhay

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May 8, 2017, 8:26:12 AM5/8/17

to BigBlueButton-dev

Hello Felipe,

I know this is an old thread I am pulling up again. but need some help. I need this native android client that doesn't work with any newer versions of BBB, hence I've to continue using the same version (0.8 beta 4) until the newer version is available.

I have made the necessary changes to the configuration files and am able to get web voice conference to work, and mobile client connects as well. While I can hear the conversations on the android client,

1) my voice from android client doesn't reach the server ( so no one else in conference can hear me )

2) in about a minute or so, the android voice conference automatically disconnects.

I have a separate thread here, could you please try and see if you could help ? Thanks much !

https://groups.google.com/forum/#!topic/bigbluebutton-dev/BFTp0DTLOEQ

Thank you

Abhay

...

abhay

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May 30, 2017, 11:43:59 AM5/30/17

to BigBlueButton-dev

Hello all, just closing the loop here.

I was finally able to get this to work on BBB 0.8 and 0.81. Still working on enabling voice conference for external SIP users on BBB 0.9.

I've also blogged about this effort on my technology blog for future wanderer :)

http://tech-voyage.blogspot.in/2017/05/extending-and-enabling-bigbluebutton.html