TrueConf Server Documentation

# Сall string formats

To make video calls or participate in conferences, you don't have to be a registered user of TrueConf Server. It is also possible to connect from any SIP, H.323 or RTSP endpoint. For each type of supported third-party protocols there is a specific call string format to be used.

It is also possible to invite users by email in which case they will receive an invitation with a link to the conference page.

A call string is a very powerful tool available in TrueConf applications. It can be used to:

  • 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 from your video conferencing server, enter his/her TrueConf ID as a call string.

You can also call a user from a different TrueConf Server instance (only if the federation has been configured between the servers). To do so use the following call string 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 of joining the conference is to use TrueConf client application: just paste the link into the search field and click the call button.

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:

  • \c\<CID>@<server>#vcs if the conference is being hosted on a different TrueConf Server instance federated with your own server. In this case:

    • <CID> is a conference ID

    • <server> is the DNS name of the server hosting the conference.

# Calling SIP endpoints

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.

You can also call the number <number> using the dialer.

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.com

  • #sip:james78@sip.company.com:5070

  • #sip:james78@sip.company.com:5070;transport=tcp

  • #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

# Calling mobile phones and landlines

You can call a phone number using the dialer in the address book. For more information about this feature, please read the TrueConf client applications user guide.

# Calling H.323 endpoints

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

  • #h323:hdx8000@192.168.1.99

  • #h323:@[fe80::805a:1cf9:12f9:def7]

  • #h323:james78

  • #h323:\e\8001

# Calling RTSP endpoints

To display an RTSP stream in the layout, add the video source as a participant to a group conference or a point-to-point call using the RTSP call string. In this way, you can access the video from an IP camera or another conference streamed over RTSP. The call string format may differ depending on the vendor or camera model. You need to check the call string format specifically for your device.

Examples of RTSP addresses for different cameras:

  • rtsp://192.168.1.100/axis-media/media.amp

  • rtsp://admin:12345scw@192.168.1.100:554/cam/realmonitor?channel=1&subtype=1

  • rtsp://admin:12345@192.168.1.100:554/Streaming/Channels/101

An example of an RTSP link for a TrueConf conference for which streaming has been enabled:

rtsp://video.server.com/c/webinar/

# Calling external users by email

If you do not know a user’s login (TrueConf ID) in the TrueConf video conferencing system and this person is not in your address book, you can call him/her by the email address. To do it, enter the call string with the prefix #mailto:, for example, #mailto:user123@example.com in the address book of your TrueConf application.

Similarly, the format #mailto:user123@example.com can be used to invite a user to a conference in the participant list menu of a client application.

# Using tone dialing

You can send DTMF commands to DTMF-compatible devices in RTP EVENT (opens new window) and SIP INFO (opens new window) modes. To learn more about the transmission of such signals, please read the documentation provided by the manufacturer for each device.

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, if you want to call a SIP server with IP 192.168.1.99 from the TrueConf client application to a conference protected by PIN 123456, you can avoid manual PIN entry by using a URI with a preset:

#sip:@192.168.1.99;123456

To call 13478783263 with extension 222, you can use the following call line:

#tel:13478783263,222

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