Сall string formats
Call String Formats
TrueConf Server enables connections in video calls and conferences not only for server users but also for SIP, H.323, and RTSP devices. Special call string formats are provided for different device types, which uniquely identify the subscriber to be contacted on the server.It is also possible to invite users by email in which case they will receive an invitation with a link to the conference page.
Calls via external SIP/H.323/RTSP protocols require licenses that should be available on the video conferencing server. If you are unable to reach a destination, please contact your administrator as this issue may be related to licensing.
Call string is an essential tool widely used in TrueConf applications. The call string formats can be utilized:
Search for a contact in a client application
Call a user from a client application
Save a new contact in the address book
Add a new participant to a conference
Create an alias
And much more.
Calling a TrueConf Server user
To call a user on your server, specify their TrueConf ID as the dial string.
A subscriber can also be a user from another TrueConf Server server (only if federation between servers is configured). To do this, use the calling line format <TrueConf_ID>@<server>, where:
<TrueConf_ID>is a user ID<server>is an IP address or a domain name of a TrueConf Server instance.
Connecting to a conference
If you have a link to the conference page, the easiest way to join it from the TrueConf client application is to paste the link into the search bar and click the call button. You can join a public conference directly from the application's start page. For more details, refer to the documentation.
It is also possible to use a call string in the following format:
\c\<CID>if the conference is being hosted on your video conferencing server, where:<CID>— conference ID;
\c\<CID>@<server>#vcsif the conference is hosted on another TrueConf Server server and federation is configured between the servers, where:<CID>is a conference ID<server>is the DNS name of the server hosting the conference.
SIP Device Call Formats
Use one of the following formats to call an SIP endpoint:
#sip:<user_id>@<server_name>, where:<server_name>is a host name or IPv4 address of an SIP server<user_id>is an SIP username.
#sip:<user_id>@[<server_name>], where:<server_name>is the IPv6 address of an SIP server;<user_id>is an SIP username.
#sip:<user_id>, where:<user_id>is an SIP username
#sip:@<endpoint_ip>, where:<endpoint_ip>is the IPv4 address of an SIP endpoint.
#sip:@[<endpoint_ip>], where:<endpoint_ip>is the IPv6 address of an SIP endpoint.
#sip:@<hostname>, where:<hostname>is the DNS name of an SIP endpoint.
#tel:<number>, where:<number>is an SIP username.
A similar call will be made when dialing from the keypad to the number <number>.
If the SIP server IP address or name is provided, you may have to specify the following parameters explicitly:
Connection port
<port>(in case it is different from the standard 5060 port)Transport protocol
<protocol>used for sending media streams (UDP is selected by default).
In such a case these parameters will be specified after the server address in the following way: :<port>;transport=<protocol>.
Call string examples for SIP protocol:
#sip:james78@sip.company.comsip:james78@sip.company.com:5070(the port is specified explicitly because it differs from the default 5060)sip:james78@sip.company.com:5070;transport=tcp(the port and TCP transport are explicitly specified because they differ from the default ones)#sip:james78#sip:8001#sip:@192.168.1.99#sip:@192.168.1.99;transport=tcp#sip:@[fe80::805a:1cf9:12f9:def7]#tel:501#tel:13478783263
Phone Calls
You can make a call to a phone number using the dialer in the client application. For more detailed information about this feature, you can refer to the client applications user guideTrueConf.
H.323 Device Call Formats
Use the following call string formats for calling an H.323 endpoint:
#h323:@<IP>, where:<IP>is the IP address of an H.323 gatekeeper.
#h323:@[<IP>], where:<IP>is the IPv6 address of an H.323 gatekeeper
#h323:<user_id>@<IP>, where:<IP>is the IP address of an H.323 gatekeeper or an MCU<user_id>is an ID of a user or a device registered on an H.323 gatekeeper with an IP address specified in<IP>parameter.
#h323:<user_id>@[<IP>], where:<IP>is the IPv6 address of an H.323 gatekeeper or an MCU<user_id>is an ID of a user or a device registered on an H.323 gatekeeper with an IP address specified in<IP>parameter.
#h323:\e\<e164_id>@<IP>, where:<IP>is the IP address of an H.323 gatekeeper or an MCU<e164_id>is an E.164 format number of a user or device registered on an H.323 gatekeeper with an IP address specified in<IP>parameter.
#h323:<user_id>@<IP>, where:<IP>is the IP address of an H.323 gatekeeper.
#h323:\e\<e164_id>@<IP>, where:<e164_id>— is an E.164 format number of an H.323 gatekeeper.
If the IP address of the H.323 gatekeeper or MCU is included, it may be necessary to specify the connection port <port> in an explicit way (when this port is different from the standard 1720 port). In this case it has to be specified after the IP address in the following way:
#h323:<user_id>@<IP>:<port>
Call string examples for H.323 protocol:
#h323:@192.168.1.99#h323:@192.168.1.99:1730(the port is specified explicitly because it differs from the default 1720)#h323:hdx8000@192.168.1.99#h323:@[fe80::805a:1cf9:12f9:def7]#h323:james78#h323:\e\8001
RTSP Device Call Format
To display an RTSP stream, add its source as a participant in a conference or one-on-one video call using the RTSP call string. For instance, this allows you to receive an image from an IP camera or another conference being streamed via RTSP. The string format may vary depending on the manufacturer and model of the camera, so please check the specifics for your device.
Examples of RTSP addresses for different cameras:
rtsp://192.168.1.100/axis-media/media.amprtsp://admin:12345scw@192.168.1.100:554/cam/realmonitor?channel=1&subtype=1rtsp://admin:12345@192.168.1.100:554/Streaming/Channels/101
Example of an RTSP link for conference TrueConf with broadcasting enabled:
rtsp://video.server.com/c/webinar/
Calling external users by email
If you don't know the user's login in the TrueConf (TrueConf ID) video communication system and they are not in your address book, you can call them using their email address. In this case, you need to specify the call string with the #mailto: prefix in the TrueConf application's address book, for example, #mailto:user123@example.com.
The email address can also be used for inviting a person to a conference, but only to a public one (webinar). You can do it in one of these ways:
pre-add them to the invitee list on the Participants tab when creating/editing a conference;
during the webinar via the participant list in the application or in real-time meeting management.
Using Dual-Tone Multi-Frequency (DTMF)
You can send DTMF commands to compatible devices using the RTP EVENT and SIP INFO modes. For more detailed information about transmitting these signals, refer to the documentation provided by the manufacturer for the specific model.
The following symbols can be used to add pauses directly to the call string:
,— short pause (a few seconds);— long pause (waiting for a dial tone from the caller).
For example, to call from the TrueConf client application to a SIP server with the IP 192.168.1.99 in a conference protected by the PIN code 123456, you can use a URI with a dial string to avoid manually entering the PIN when connecting:
#sip:@192.168.1.99;123456
To call 13478783263 with extension 222, you can use the following call line:
#tel:13478783263,222
You can also send DTMF commands from endpoints during a moderated role-based conference. The following DTMF commands are available:
1– request for the podium2– leave the podium.
To do it, use the remote control included in the kit or keyboard. For more details, refer to the instructions for your specific device.