SIP/H.323/RTSP Gateway and Transcoding

TrueConfAbout 39 min

SIP/H.323/RTSP gateway and transcoding

TrueConf Server It includes a built-in gateway for SIP 2.0, H.323, and RTSP protocols, which can be configured in the Gateways section of the control panel.

With the gateway you can:

Built-in gateway is necessary only if you need to call the devices connected to a third-party server (e.g. H.323 gatekeeper, PBX, MCU). Otherwise you can use the call string for SIP 2.0/H.323 devices.

If your PBX and TrueConf Server are integrated with an LDAP directory (such as Microsoft Active Directory), additional fields can be used for convenient user dialing. For instance, you can retrieve aliases from LDAP and use them for calling. For more details, read the LDAP fields description.

SIP Gateway

This section helps to configure TrueConf Server built-in SIP 2.0 gateway parameters. The number of rules created using these settings is unlimited.

TrueConf Server Free version provides only one active connection through the gateway, including SIP 2.0, H.323 and RTSP protocols.

To call devices through the SIP gateway in TrueConf Server, a special call string format is provided.

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Block "Network Settings"

This list contains the addresses where the gateway will await incoming SIP 2.0 connections. By default, it is pre-filled with IP addresses provided by the operating system. To edit the list, uncheck the box Listen on all IP addresses.

Section "Rules for SIP Connections"

In this section you can create specific rules for certain SIP addresses or call directions. For example, you can use special set of settings to connect to Skype for business servers and another one for PBX connectivity. Every rule is relevant only for target address specified in Host field. Every rule redefines global settings for SIP 2.0 connections.

Gateway can also authenticate on and maintain active connection with SIP devices for which the rules have been created. This option can be useful to maintain permanent connection with PBX or VoIP services. You can find the connection status in the rules for SIP Connections table.

To create a new rule, click the Add a configuration button. You will be prompted to choose from two templates: manual setup and server connection setup Skype for business. The Skype for business template sets certain properties needed for its correct operation, such as the port in use, protocol, video codec, and registration mode.

Form for creating a new rule

The first group of settings affects the forwarding of SIP connections and authorization (if needed):

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Name field is only displayed in the table for rules. Host and Port fields are more important and also mandatory. They are required to determine call direction applied to this rule. If you are using an SIP proxy server, enter its IP address or domain name in the corresponding field. If the port for connecting to the proxy is different from the 5060 default port, enter the required port after the address and separate it with a colon. Please note that it isn't possible to set different rules for one host but different ports.

In the External NAT IP address field, you can specify the server's IP address to be listed in the SDP for receiving and sending media streams when calling users behind NAT.

The field Outgoing SIP domain for callback to TrueConf Server is used to create a SIP URI in the format user@server for outgoing calls to SIP devices, where server is the entered IP address or FQDN, and user is replaced with the ID of the user initiating the call. It is typically displayed as the caller's address on SIP devices. Possible values:

  • Do not specify, then only TrueConf ID will be in the address;
  • Use server public name, then the external server address specified in section Web → Settings is substituted;
  • Use other domain, specifying the desired domain name in the input field.

The next set of fields is intended for configuring authentication on a SIP device for which a rule is being created. If Authorization name is the same as the login, it can be left blank. Specifying International call prefix allows you to replace the + symbol, which users input when calling phone subscribers, with another value, such as 810. If this field is left empty, the + in phone numbers will not be replaced.

The Registration mode determines how registration will be conducted in this direction:

  • off — the REGISTER request is not sent, and registration or authorization on the external SIP device does not occur;
  • permanent — registration occurs automatically each time TrueConf Server is launched;
  • before call — Registration occurs immediately before each call and remains active only during the call.

You can manually specify the connection protocol (TCP, UDP or TLS) if necessary.

Please note that each active gateway connection reserves one SIP 2.0/H.323 connection from TrueConf Server license.

Next, you can find the settings for the transfer of data and other advanced parameters:

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Use block Reduce SIP messages size to compress SIP message packets and headers, preventing issues related to exceeding their maximum allowable size (MTU).

The checkbox Enable ICE support (Interactive Connectivity Establishment) determines whether the gateway will be available if TrueConf Server is located behind NAT.

When the checkbox Enable SRTP support is enabled, it ensures the encryption of media data transmitted in this direction. Some SIP devices require this (for example, Skype for business servers).

The checkbox Enable forward error correction (FEC) enables control over error correction when the connection deteriorates on a configurable SIP route. It is enabled by default, but some devices or server MCUs may not function correctly with this setting, in which case it should be disabled. If you are configuring rules for connecting to TrueConf Group or TrueConf MCU, we recommend keeping the checkbox Enable forward error correction (FEC) enabled.

The checkbox Enable content sharing via BFCP determines whether the server can exchange content sharing with SIP devices in this direction by transmitting content as an additional video stream. For example, this could be used to share the desktop screen from a computer connected to a SIP terminal or to send slides from applications TrueConf to a SIP terminal.

When content is shared from SIP/H.323 devices in the secondary stream, it is sent with a reduced frame rate to reduce traffic (similar to the transmission of the secondary stream from TrueConf client applications).

The checkbox Enable far end camera control via Q.922/H.224/H.281 determines whether remote control of SIP device cameras will be available from the TrueConf client application.

Please note that this parameter has the same name in the SIP and H.323 gateway configuration menus, however, these are two different checkboxes responsible for different permissions.

The checkbox Enable timers support (RFC4028) is used to disconnect the SIP terminal from the conference if the connection is lost. It is disabled by default.

You can manually set Max session refresh interval (seconds) (default is 1800 seconds).

The list of Available codecs contains codecs that the gateway can use in this direction. Disabling some codecs may resolve compatibility issues with certain SIP devices. You can learn more about this from our technical support service.

SIP device for which the rule is created can take special roles:

  • Default SIP trunk . This role allows users to avoid entering full SIP URI for calls with #sip: prefix. For example, all calls in the #sip:Endpoint format will be automatically replaced with #sip:Endpoint@Host, where Host is taken from the properties of this rule and Endpoint is a username specified during the call.
  • Default VoIP server . This role is required for treating an SIP device as a VoIP server or a PBX and activating the dialers built in TrueConf client applications. All the calls made from application dialers or with the help of #tel: prefix will be automatically forwarded to this SIP endpoint. For example, #tel:Phone will be automatically replaced with #sip:Phone@Host, where Host parameter is automatically taken from the properties of this rule and Phone is replaced with the phone number entered by user.

Please note that each of these roles can be assigned only for one SIP 2.0/H.323 connection rule.

Setting up integration with Skype for Business

This integration is designed to work with Skype for business 2015 Server or Lync 2013 Server on-premises deployments and cannot be used for their cloud versions.

To connect successfully, you will need to receive a trusted root certificate from the Skype for business administrator and install it in the system where TrueConf Server is installed.

  1. Create a new account for TrueConf Server on the Skype for business server.

  2. Use Skype for business template to create a new rule for SIP connections. Enter username and password of this freshly created account in the appropriate fields.

  3. In the Host field, enter the IP address or domain name of the Skype for business server.

  4. Check the box Default SIP proxy.

  5. Save the rule and check if the connection status has changed to successful in the table for rules. Please note that TrueConf Server service must be running.

To call Skype for business subscribers from TrueConf client applications, use the format #sip:<user>, where <user> is the login of the Skype for business user. An incoming call to this user will be displayed as coming from the account created for TrueConf Server. Similarly, Skype for business subscribers can be invited to conferences or added to the address book.

To call TrueConf users from Skype for business client application, send the following message to the user created for TrueConf Server authentication: /call <USER_ID>, where <USER_ID> is any valid TrueConf Server user ID including SIP / H.323 devices registered on TrueConf Server. You can use /conf command to create a multipoint conference, etc. After the message has been sent, TrueConf Server will call Skype for business user and connect him/her to a TrueConf user or a conference. If you try to call this user directly, the call will be rejected and you will receive a help message with a list of available commands in chat. However, if default call destination is set in global SIP settings, you will be connected to this default destination address.

Please note that you can also create a group conference on TrueConf Server and invite into the conference the endpoints connected via any protocols the gateway supports. For example Skype for business users and various SIP/H.323 devices or RTSP IP cameras.

Global SIP settings

Settings in this section automatically apply for all SIP 2.0 connections for which there are no rules.

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Action on incoming call to TrueConf Server IP address — this parameter allows you to select the behavior for a SIP call to any of the addresses from block Network settings using the SIP 2.0 protocol:

  • automatically reject such a call;

  • transfer the call to the conference ID input menu using DTMF;

  • Transfer the call to TrueConf ID user or conference ID. You should then enter this ID in the field below.

The remaining settings are identical to those used when creating rules for connections.

Invitation of the SIP endpoint to the conference on your server

There are several ways to invite a SIP terminal to a conference: the conference owner can call the terminal during the conference using the client application TrueConf, employing a special call string format, or an administrator can add the terminal to the conference from the server control panel.

To add an SIP endpoint to the conference via control panel you need to:

  • select a conference on the conference list page;

  • Add the SIP terminal as a conference participant if it hasn't started yet, or invite it to an ongoing conference. To do this, use the call string format.

How to join a conference with its CID (conference ID) from an SIP endpoint

To connect to a conference from the endpoint registered on TrueConf Server, enter CID (Conference ID) into the endpoint address field. Please note that you need to replace \c\ in CID with 00 (two zeroes) when calling from external endpoints. In our case, you need to enter 00e22a39ba2a@<server> if CID is equal to \c\e22a39ba2a.

To connect to the conference from the endpoint unregistered on TrueConf Server, use the following format:

CID@<server>:<port>

where:

  • CID is a conference ID with two leading zeroes instead \c

  • <server> — the IP address of the TrueConf Server gateway (for example, 00e22a39ba2a@192.168.1.99);

  • <port> — connection port (in case it is different from the standard 5060 port).

Additionally, in the case of SIP it is possible to specify the protocol name explicitly (UDP is used by default):

CID@<server>:<port>;transport=<protocol>

For example, 00e22a39ba2a@192.168.1.99:5061;transport=TCP.

You can also find an instruction on how to connect to a conference held on TrueConf Server from an SIP endpoint on the conference web page.

H.323 Gateway

This section explains how to configure built-in gateway parameters for H.323 connections. The number of rules for H.323 connections created using this section of control panel is unlimited.

TrueConf Server Free version provides only one active connection through the gateway, including SIP, H.323 and RTSP protocols.

Connections using the H.323 protocol are primarily used for calls to third-party hardware video conferencing terminals. TrueConf Server also allows configuring integration with MCU, H.323 gatekeeper, and PBX. This can be useful for addressing devices and subscribers registered on them using H323-ID or E.164 format numbers without specifying the terminal's IP address in the call string. For calling devices via an H.323 gateway in TrueConf Server, a special call string format is provided.

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Block "Network Settings"

This section contains a list of addresses where the gateway will wait for incoming H.323 connections. By default, it is pre-filled with the IP addresses of the operating system. You can edit the list by unchecking Listen on all IP addresses. The list of ports used for H.323 connections can be found in our blog article.

Section "Rules for H.323 Connections"

Here you can create specific rules for certain H.323 devices or call directions. Each rule is relevant only for specific destination address indicated in the Host field and redefines global settings for H.323 connections.

The gateway can also register on H.323 devices and maintain an active connection, which might be useful when connecting to an MCU or H.323 gatekeeper. The status for such connection is displayed in the rules table. To create a new rule, click Add a configuration button.

Form for creating a new rule

The field Name is used only for display in the rule list. Host and Port are also required and are used to determine the direction of calls to which this rule will apply. Note that it is not possible to create two rules with the same host but different ports.

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In the External NAT IP address field, you can specify the server's IP address to be listed in the SDP for receiving and sending media streams when calling users behind NAT.

The fields H323-ID and Password are meant for authentication on the H.323 device for which the rule is created. To maintain a constant connection with the device, you need to select the appropriate option from the drop-down list Registration.

After successful registration, the H.323 device TrueConf Server will also be available for dialing using the E.164 number format, provided it was specified in the DialedDigit field. This feature is beneficial when used in conjunction with the Default call destination field in the global H.323 settings. In this case, all calls to this DialedDigit number originating from a connected H.323 device will be redirected to a specific user or conference on TrueConf Server.

Please note that each active gateway connection reserves one SIP/H.323 connection from TrueConf Server license.

The checkbox Enable H.235 encryption is designed to enable encryption of media data transmitted to H.323 devices according to ITU-T H.235 version 3, which is necessary for the proper functioning of certain terminals.

Enabling the checkbox Enable content sharing over H.239 allows the sending and receiving of content from an H.323 device as an additional video stream. For example, it can be used to share the desktop screen from a computer connected to an H.323 terminal or to send slides from TrueConf applications in the reverse direction.

When content is shared from SIP/H.323 devices in the secondary stream, it is sent with a reduced frame rate to reduce traffic (similar to the transmission of the secondary stream from TrueConf client applications).

The checkbox Enable far end camera control over Q.922/H.224/H.281 determines whether remote camera control via Q.922, H.224, H.281 protocols will be available through client applications TrueConf.

Please note that this parameter has the same name in the SIP and H.323 gateway configuration menus, however, these are two different checkboxes responsible for different permissions.

The list of Available codecs displays the codecs which gateway is allowed to use in this direction. Disabling some of the codecs can solve compatibility issues with certain H.323 devices.

H.323 device for which the rule is created can take special roles:

  • Default H.323 gatekeeper — allows not specifying the full address of the called device when making H.323 device calls through the #h323: prefix. For instance, all calls made by server users in any direction in the #h323:Endpoint format will automatically be replaced with #h323:Endpoint@Host, where the Host parameter is taken from the properties of this rule, and Endpoint is the username specified during the call.
  • Default VoIP server . This role is required for treating an H.323 device as a VoIP server or a PBX and activating the dialers built in TrueConf client applications. All the calls made from application dialers or with the help of #tel: prefix will be automatically directed to this H.323 endpoint. For example, #tel:Phone will be automatically replaced with #h323:Phone@Host, where Host parameter is automatically taken from the properties of this rule and Phone is replaced with the phone number entered by user.

Please note that each of these roles can be assigned only for one H.323 rule.

Global H.323 Settings

Most of the settings in this section are identical to the settings described above. However, they automatically apply for all H.323 connections for which there are no rules.

The Action on incoming call to TrueConf Server IP address parameter allows you to select the behavior for a SIP call to any of the addresses from the block Network settings using the SIP 2.0 protocol:

  • automatically reject such a call;

  • transfer the call to the conference ID input menu using DTMF;

  • Transfer the call to TrueConf ID user or conference ID. You should then enter this ID in the field below.

Methods for calling subscribers and conferences on your server using H.323 devices

Depending on the H.323 endpoint model there are two different methods to call TrueConf Server users and conferences: using SIP URI or hashes (##) notation. Please try both to find the one suitable for your H.323 equipment. The call strings provided below should be entered as a string or number to call in the endpoint’s interface. TrueConf Server IP address mentioned below could be an any address specified in H.323 network settings section:

  • Server##User, where Server is the IP address of TrueConf Server, and User is the ID of the user or device registered on TrueConf Server;

  • Server##00CID, where Server is the IP address of TrueConf Server, and CID is the conference ID on TrueConf Server;

  • User@Server, where User is ID of the user or device registered on TrueConf Server and Server is TrueConf Server IP address

  • \c\CID@Server, where CID is the conference ID on TrueConf Server, and Server is the IP address of TrueConf Server;

  • 00CID@Server, where the first two characters are zeros, CID is the conference ID on TrueConf Server, and Server is the IP address of TrueConf Server.

For more details, H.323 call formats examples are described in the user documentation.

Registering H.323 devices on the video conferencing server

TrueConf Server can act as a gatekeeper or MCU for third-party H.323 devices and simplify their addressing. From the TrueConf Server user perspective an H.323 device registered on the server does not differ from any other user: you can see its status, call it from the address book or invite to the conference without using call strings notation. Similarly, calls using H323-ID names from a registered H.323 device interface will be interpreted by the server as a call to specific TrueConf ID to entered H323-ID.

Registering an H.323 device on TrueConf Server is similar for most endpoints available on the market. Basically, to do so, you will need to specify TrueConf Server address as a gatekeeper or MCU address and use username and password of any TrueConf Server account to authenticate.

Sending DTMF Commands

Thanks to the ability to process DTMF signals, during a conference in the “moderated role-based conference” mode, you can send the following DTMF commands from your SIP/H.323 terminal:

  • 1 – request to take the podium.

  • 2 – to leave the podium.

To do this, use the supplied remote control or keypad. For more details, read the manuals for your specific device.

In our knowledge base, we have covered the use of TrueConf Server with Polycom HDX series terminals, including sending DTMF commands from them.

Chat during calls on MCU

When meeting participants make calls from TrueConf client applications to conferences created on TrueConf MCU, they will be able to make use of chats that work via H.323 / SIP. This means that users who have signed in to TrueConf Server are not only able to make calls to TrueConf MCU, but can also send messages. The text of such messages will overlay the video layout, and all conference participants will see it regardless of their connection method:

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RTP

In section Gateways → RTP, you can configure the UDP port range used for media exchange during SIP/H.323 calls (default is 50000-51999).

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WebRTC

In this section, one can configure parameters for connecting conference participants via WebRTC (from a browser):

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  • Set up UDP or TCP port range for WebRTC connection (the ports 53000-56000 are used by default). You can also choose the protocol for working with WebRTC: TCP, UDP or both.

  • In the Public IP address is added to SDP as an extra ICE candidate field, you can specify the IP address to be used for NAT traversal if automatic detection fails for any reason. You can choose the protocol for NAT traversal at this address: TCP, UDP, or both.

Transcoding

In the Gateways → Transcoding section, you can set the background and watermark for the video layout, as well as video quality for different types of connections and recording.

Adjusting Quality

In the section Restrictions for modules, one can configure conference video quality for WebRTC users (joining from a browser), H.323/SIP/RTSP devices, and recording. In other words, here you can set the quality of video streams outgoing from the server in these directions.

These settings do not apply to the transcoding of the second stream for SIP/H.323 participants, in other words, they do not affect content sharing via BFCP/H.239: protocols:

  • If BFCP/H.239 content is transmitted from one endpoint to another, it will be delivered in its original quality, for example, 720p @ 60 fps.

  • If participants connected through TrueConf client applications receive BFCP/H.239 content from an endpoint, the content will be delivered to them in its original resolution (up to 1080p) at 0.5 fps (1 frame every 2 seconds). This behavior is similar to when content is shown in the second stream from a client application.

The video quality settings from conference participants to TrueConf Server are selected in the conference settings.

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Enabling the checkbox Do not display self-view in video layout for H.323 and SIP endpoints allows you to create a conference video layout for each SIP and H.323 terminal without the video window of that particular device. This means that an individual video layout is generated for the SIP/H.323 participant, excluding the image from the camera connected to it.

Enabling the checkbox Do not display self-view in video layout for WebRTC participants will create a conference video layout for each browser connection without including the participant's own video window. This means an individual layout is created for the WebRTC connection, excluding the browser's camera feed.

The checkbox Do not display content from the second stream to SIP/H.323/WebRTC participants allows you to exclude the mixing of the second stream (which transmits content or a slideshow) from the final video layout for all SIP/H.323/WebRTC connections to conferences. However, if the server-side video recording of the conference is enabled, a separate mixing process will be initiated, and the second stream will be included in the recording.

Please note that if a video layout is set at the conference level (either in advance or in the real-time meeting management tool during the session), the settings for excluding the self-view (which were discussed earlier) will no longer apply to the current conference. In other words, the settings of a specific conference have priority over general settings.

By creating individual video layouts for each SIP/H.323 and WebRTC endpoint or excluding the content stream, you can significantly increase CPU load on the physical machine where TrueConf Server is installed.

If the box Use GPU to reduce CPU load is checked, conference video will be processed by the GPU of the physical machine where the server is installed. NVIDIA graphics cards with CUDA support are suitable, preferably server versions (such as Quadro P2000). The graphics card will process the video of individual layouts which are generated for SIP/H.323/WebRTC protocols according to the settings discussed earlier:

  • Do not display self-view in video layout for H.323 and SIP endpoints;
  • Do not display self-view in video layout for WebRTC participants;
  • Do not display content from the second stream to SIP/H.323/WebRTC participants.

GPU can be used for transcoding only in TrueConf Server for Windows. Not all models may be suitable, so please contact our technical support before estimating the possible encoding acceleration.

The parameter Automatically spotlight active speaker window activates automatic speaker window enlargement based on voice activity. Note that the settings for hiding your video in the layout and automatic speaker enlargement will only work if a layout is not explicitly set for SIP/H.323/WebRTC participants during conference scheduling or in real-time meeting management.

Adding Background and Watermark

In section Gateways → Transcoding → Visual settings, you can select the global background and watermark settings for the video layout of all conferences. Once you choose a watermark image, you can specify its position in the layout.

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Adding a watermark is not available in version TrueConf Server Free; a paid license is required.