This is documentation for Web Call Server 3.x

Take a look at our newest product Web Call Server 5

Web Call Server 5 product page

Configuring


Client configs

flashphoner.xml

Located on your web server at: /flashphoner_client/flashphoner.xml

flashphoner.xml
<flashphoner>
        <wcs_server>192.168.1.5</wcs_server>
        <ws_port>8080</ws_port>
	<wss_port>8443</wss_port>
	<use_wss>false</use_wss>
        <rtmfp_port>1935</rtmfp_port>
        <register_required>true</register_required>
        <!--<token>etgrtgrth</token>-->
        <application>phone_app</application>
        <video_width>176</video_width>
        <video_height>144</video_height>
        <check_validation_callee>true</check_validation_callee>
        <use_enhanced_mic>true</use_enhanced_mic>
        <ring_sound></ring_sound>
        <busy_sound></busy_sound>
        <register_sound></register_sound>
        <finish_sound></finish_sound>
</flashphoner>
Parameter name Default value Format Description
<wcs_server> 192.168.1.5 String Address of Flashphoner server to which you want to connect
<ws_port> 8080 int Websocket port
<wss_port> 8443 int Websocket secure port
<use_wss> false Boolean If true, WCS client will try to connect using Websocket secure protocol to WCS server. Works only with WebRTC.
<rtmfp_port> 1935 int RTMFP port
<register_required> true Boolean Is register on the VoIP server required? Set it to false if you will use Flashphoner client as a click2call button
<load_balancer_url>
Empty String Specifies URL of load balancer server, for example http://192.168.1.5:8081/?action=server_list
<token> Commented String Uncomment and set the token to enable the AutoLogin feature.
<application> phone_app String Server-side application name, phone_app by default. This application is bundled with Flashphoner.
<video_width> 176 int Maximum video width of outgoing video stream Flash to SIP. Used for H.264 codec only.
<video_height> 144 int Maximum video height of outgoing video stream Flash to SIP. Used for H.264 codec only.
<push_log> false Boolean Enable push of client log report to WCS server.
<check_validation_callee> true Boolean Should Flashphoner client validate input data (like phone number or SIP account name)?
<use_enhanced_mic> true Boolean Use enchanced microphone with echo cancellation feature. See also Echo cancellation
<ring_sound> - File name File name for ring sound. Leave it blank to use Flashphoner default
<busy_sound> - File name File name for busy sound. Leave it blank to use Flashphoner default
<register_sound> - File name
File name for register sound. Leave it blank to use Flashphoner default
<finish_sound> - File name
File name for finish sound. Leave it blank to use Flashphoner default
<msg_content_type> text/plain String text/plain, message/cpim types are supported
<stun_server> Empty String Custom stun server address, used for WebRTC.
<xcap_url> Empty String Send XCAP request. %username parameter can be included. For example: http://domain.com/page_%username?id=1
<msrp_callee> Empty String Sip Uri Initiate MSRP call upon REGISTERED event to the callee.
<subscribe_event> Empty String reg. event is supported. Subscribe request will be sent upon REGISTERED event. Example: <subscribe_event>reg</subscribe_event>
<contact_params> Empty String Will be added to REGISTER request Contact Header params. Example: q=1.0;a=2;b=3
<imdn_enabled> False Boolean Enable IMDN for incoming and outgoing instant messages
<strip_codecs> Empty String Comma-separated list of codecs which should be excluded from WebRTC SDP
<streaming> Empty String Supported values: webrtc, flash. If set to either, WCS client uses specified technology. This setting is useful for developers.

Server configs

flashphoner.properties

Located in: /usr/local/FlashphonerWebCallServer/conf/flashphoner.properties
Critical parameters marked red

Parameter name Default value Format Description
ip - xxx.xxx.xxx.xxx External IP-address of server where Flashphoner is installed more info
ip_local - xxx.xxx.xxx.xxx Local IP-address of server where Flashphoner is installed more info
Ports      
port_from 30000 Int (1, 65536) Beginning of range of ports for SIP signaling
port_to 31000 Int (1, 65536) End of range of ports for SIP signaling
media_port_from 31001 Int (1, 65536) Beginning of range of ports for media-traffic
media_port_to 32000 Int (1, 65536) End of range of ports for media-traffic
License      
serial_number - xxxxxxxxxxxxxxxxxxxxxxxxxxxxxxx... Used only for WCS version 2.1. For more info on licensing, see License Activation section.
Headers      
user_agent Flashphoner/1.0 Flashphoner/1.0 SIP User Agent
balance_header balance String SIP header name for balance info
cost_header cost String SIP header name for cost info
SIP provider      
domain - xxxxxxx.xxx / xxx.xxx.xxx.xxx Domain address (domain name or IP address). If you specify it here - the parameters sip_proxy:port from the client will be ignored. The above will be used for all webphones.
If you use the auto login model, please be careful - domain:port from account.xml config will be dominated by domain:port from flashphoner.properties config.
outbound_proxy - xxxxxxx.xxx / xxx.xxx.xxx.xxx Outbound proxy (domain name or IP address). If you specify it here - the parameters sip_proxy:port from the client will be ignored. The above will be used for all webphones.
If you use the auto login model, please be careful - domain:port from account.xml config will be dominated by domain:port from flashphoner.properties config.
outbound_port - Integer Outbound port. Used for connecting ports different from the default 5060.
Audio / Video      
video_enabled
true
Boolean
Is video support enabled?
profiles 42001e,42000a
String
H.264 profiles supported by Flashphoner
rtp_activity_detecting true,5 Boolean,int Is RTP actitvity detecting enabled?
If true - calls will be terminated when one side stops sending RTP packets
int - amount of time in seconds, after which call is terminated
codecs speex16,alaw,ulaw,telephone-event,g729,flv,h264,h263 codec1,codec2,codec3... Comma separated list of codecs ordered by priority. Available values: alaw, ulaw, speex16, g729, h263,h264,telephone-event  
speex16 must be included in codec list because Flash Player uses this codec

video_resolution_detecting
true
Boolean
If true, notify client about video resolution of incoming H.264 stream. If false, use default resolution 176x144
record
- String Folder where you would like to save recorded audio calls. For example: /tmp/mycalls
If this parameter is empty, calls will not be recorded.
See the Call Recording page for more info.
preserve_non_mixed_recorded_files
false Boolean Keep "incoming" and "outgoing" recorded files if true. If this value is false, Flashphoner will keep the mixed result file only.
Signaling      
dtmf
rfc2833
String Set DTMF signal format. Vaules: info, rfc2833, info_relay
sip_msg_listener - <full java class name> Used for intercepting and overriding SIP messages between Flashphoner and VoIP servers 
See our page about ISipMessageListener interface
send_busy_when_on_call
false Boolean Send BUSY if we have an established call between two users and third user is calling to one of these users.
QoS and TOS      
sip_traffic_class 0 int [0,255] Type of service value for outgoing SIP traffic. See QoS and TOS support for more details.
rtp_traffic_class 0 int [0,255] Type of service value for outgoing RTP traffic. See QoS and TOS support for more details.
Quality / latency tuning      
speex_in_policy good Integer Speex Voice Quality Tuning
other      
options2flash_delegate
false Boolean Used for transferring OPTIONS requests to the Flash client.

Click2call related parameters      
get_callee_url - String HTTP URL that points to a dynamically created XML file, see Click2call configuration
auto_login_url
- String HTTP URL that points to a dynamically created XML file, see Click2call configuration


options2flash_delegate=true
Server will check if the Flash client is still connected using OPTIONS requests, and if so - send OK as an answer.
options2flash_delegate=false
Server will immediately answer OK using incoming OPTIONS requests.

server.properties

Located in: /usr/local/FlashphonerWebCallServer/conf/server.properties

Parameter name Default value Format Description
keep_alive.algorithm
HIGH_LEVEL
Enum, Value Range Can be INTERNAL, NONE, HIGH_LEVEL. Server side 'keep alive' algorithm.
INTERNAL - protocol-level 'keep alive' for Websocket (ping pong) and RTMFP (keep alive) protocols will be used
HIGH_LEVEL - we will only send ping pong messages over the underling protocol Websocket or RTMFP and control connections using custom 'keep alives'
NONE - do not use server-side 'keep alives' at all
keep_alive.peer_interval
2000
Int 'Keep alive' peer interval in milliseconds
keep_alive.server_interval
5000
Int 'Keep alive' server interval in milliseconds
keep_alive.probes
5 Int Amount of unanswered keep alives before client is disconnected
video_reliable
partial on, partial, off Setting of RTMFP protocol reliability for video streams. See also NetStream AS3 Class Reference.
audio_reliable partial on, partial, off Setting of RTMFP protocol reliability for audio streams
http_port 8081 Int This port will be used to return statistics by URL http://ip:8081?action=stat
ws.port 8080 Int Websocket port
wss.port 8443 Int SSL websocket port
wss.keystore.file conf/wss.jks String Java keystore file path which contains certificates for secure websocket connections
wss.keystore.password password String Password for wss.keystore.file
audio_frames_per_packet 6 Int Pack audio frames into one UDP packet to decrease bandwidth
burst_avoidance_count 100 Int Threshold value. When this threshold value is exceeded, RTMFP pauses the sending of packeets
flush_audio_interval 80 Int Number of milliseconds. Interval of flushing packets from internal queue to RTMFP downstream
flush_video_interval 0 Int Number of milliseconds. Interval of flushing packets from internal queue to RTMFP downstream
load_balancing_acao_header Empty string String Additional HTTP ACAO header, when we provide load balancing information over HTTP protocol
stream_mode_udp true Boolean true, false Use an RTMFP hack to avoid TCP-like retransmissions and make RTMFP audio or video upstream similar to plain UDP

Logs

Tcpdump logs

  1. Install Tcpdump
    $yum install tcpdump
  2. Run tcpdump logging
    $tcpdump udp -s 2048 -w flashphoner.pcap
  3. Stop tcpdump logging
    Ctrl+C

Stdout logs

  1. Edit FlashphonerWebCallServer/bin/startup.sh file. Replace:
    /dev/null

    with

    $APP_HOME/logs/flashphoner_stdout.log

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