We announced a new communications product, Hangouts, in May 2013. Hangouts will replace Google Talk and does not support XMPP.

Developer Scenarios

You can modify libjingle code to support a number of different scenarios:

Scenario Description
Video chatting To create a video chat application, start with the call sample code and add a media engine to render and capture video. If you are using Jingle, use the Jingle video media description format (XEP-0180).
Music streaming Streaming live music could be done either by modifying the call sample code to enable one-way communication only, or by using the File Share sample application and replacing FileStream with MemoryStream. You would have to modify the code to set one computer as the server, and handle the session negotiation to determine which computer is serving, similar to how the file share sample application specifies the serving computer. (Part of this would include modifying the SessionDescription to include custom data to indicate the server). You could replace the voice engines with the basic audio renderers on the operating system.
Tuning the port allocation You might want to extend the HttpPortAllocator class to specify how relay resources are allocated. You could also modify the Transport class to change the candidate prioritization algorithm.
Supporting a different signaling protocol Although libjingle uses XMPP as its session negotiation protocol, you could modify the code to support a different protocol. If you use an XML-based protocol you could reuse most of the existing code but change the SessionDescription object and modify the PhoneSessionClient::TranslateSessionDescription methods, and provide your own QName values to describe the elements and attributes of your protocol. For non-XML based protocols you would have to provide a more extensive reworking of the XMPP messaging component, Session, SessionClient, and SessionManager classes.

 Most of these scenarios involve the following two tasks:

Supporting Additional Session Types

libjingle provides code to support voice chat applications and file transfers. If you want to support additional types of data transfer, you will have to extend a number of base classes, and possibly provide new rendering, wrapping, or helper classes. The following list describes the steps to support an additional session type.

Extending SessionClient

The SessionClient class manages all session objects of a specific type for the application. It handles tasks that are common across all sessions, such as specifying which connection should be active, accepting or rejecting a session request, and initiating connection requests. Extensions of the SessionClient class typically create any additional helper classes you might need, such as media engines or stream wrappers. You must extend SessionClient and SessionDescription in order to handle new connection types.

When extending SessionClient, you should override the following virtual methods as described.

SessionClient Method Description
OnSessionCreate Called by the SessionManager object when it creates a new Session object for either an incoming our outgoing connection request. In the implementation, connect to the Session object's Session::SignalState and Session::SignalError signals. Your handler method should parse the session-specific description passed in for information, such as codecs, and take the appropriate action.
OnSessionDestroy Called by the SessionManager when it destroys a Session object. Your implementation should clean up all resources allocated for the Session object and links to the Session object created by your SessionClient implementation.
CreateSessionDescription Translates an incoming XMPP stanza describing the specifics of the session into a SessionDescription subclass. This is described in the following section.
TranslateSessonDescription Converts a SessionDescription object sent by Session into an XMPP stanza ready for transmission across the network.

Extend SessionDescription

When negotiating a connection between two clients, you may need to include additional information in the stanzas sent between the two computers. For instance, when setting up an audio session, both computers must exchange a list of audio codecs supported by each application. Some protocols have requirements about what must be included (for example, audio sessions exchange audio codec lists), but you can include any additional information that is understood by both parties in the connection. This additional information is wrapped by the SessionDescription class when translated from the XMPP stanza.

Extend SessionDescription to include any kinds of methods and variables you need to describe your session. Both the Session and SessionClient objects must be customized to be able to parse and use the additional information. The SessionClient subclass is responsible for translating between the stanza and the SessionDescription class in its implementation of CreateSessionDescription and TranslateSessionDescription.

Translating a SessionDescription object to an XmlElement object

libjingle uses the XmlElement object to wrap an XMPP stanza. The following methods and describe how to build an XmlElement from a SessionDescription object.

To create a stanza encapsulating a session description, you must create an XmlElement object with the name "description" and a namespace unique to your session type. The XmlElement object constructor takes both of these parameters wrapped in a QName object as shown here (taken from PhoneSessionClient):

const std::string NS_PHONE("http://www.google.com/session/phone");
const buzz::QName QN_PHONE_DESCRIPTION(true, NS_PHONE, "description");
buzz::XmlElement* description = new buzz::XmlElement(QN_PHONE_DESCRIPTION, true);

The XmlElement object defines the following constructor:

XmlElement(QName &name, bool useDefaultNs)

A QName (qualified name) object combines the element name and a namespace into a single object. QName objects in libjingle are typically global scope. The PhoneSessionClient class, for example, defines several QName objects required by Jingle Audio.
A boolean value specifying whether to qualify with the namespace when printing XML. You will always need to create an XML element with the name "description" and a namespace unique to your session type.

You may then need to add child elements to this root description element with the AddElement method. A file sharing session description might create one element for each file; PhoneSessionClient creates one for each supported codec. The following code from PhoneSessionClient demonstrates creating a SessionDescription with a sub-element for each codec:

buzz::XmlElement *PhoneSessionClient::TranslateSessionDescription(const SessionDescription *_session_desc) {
  // Cast the generic SessionDescription to the specific type needed here.
  const PhoneSessionDescription* session_desc = static_cast<const PhoneSessionDescription*>(_session_desc);
  // Create the wrapper element.
  buzz::XmlElement* description = new buzz::XmlElement(QN_PHONE_DESCRIPTION, true);

  // Loop through the codecs in the SessionDescription and add one child element to 
  // the stanza for each codec.
  for (size_t i = 0; i < session_desc->codecs().size(); ++i) {
    buzz::XmlElement* payload_type = new buzz::XmlElement(QN_PHONE_PAYLOADTYPE, true);

    char buf[32];
    sprintf(buf, "%d", session_desc->codecs()[i].id);
    payload_type->AddAttr(QN_PHONE_PAYLOADTYPE_ID, buf);



  return description;

The code shown above creates the following audio description stanza:

<description xmlns='http://www.google.com/session/phone'>
    <payload-type id='18' name='G729'/>
    <payload-type id="97" name="IPCMWB"/>
    <payload-type id='98' name='L16'/>
    <payload-type id="103" name="ISAC"/>
    <payload-type id="102" name="iLBC"/>
    <payload-type id="4" name="G723"/>
    <payload-type id="100" name="EG711U"/>
    <payload-type id="101" name="EG711A"/>
    <payload-type id="0" name="PCMU"/>
    <payload-type id="8" name="PCMA"/>
    <payload-type id="13" name="CN"/>

Translating an XmlElement object to a SessionDescription object

Reading data from an XmlElement is similarly straightforward. Your CreateSessionDescription method will be passed the received <description/> element, from which you should follow whichever protocol being used to extract data out of it. There are a few libjingle functions that should be particularly helpful.

XmlElement::FirstNamed returns the first child element with a given qualified name. PhoneSessionClient::CreateSessionDescription calls FirstNamed to get the first <payload-type/> child:

  const buzz::XmlElement* payload_type = element->FirstNamed(QN_PHONE_PAYLOADTYPE);

You can then loop through each <payload-type/> element by using the XmlElement::NextNamed method:

    payload_type = payload_type->NextNamed(QN_PHONE_PAYLOADTYPE);

The specifics of how to interpret the XMPP stanza will vary depending on the session type. For example, because Jingle Audio calls for a number of <payload-type> elements, it uses a loop to add each one to the PhoneSessionDescription object:

int id = atoi(payload_type->Attr(QN_PHONE_PAYLOADTYPE_ID).c_str());
  int pref = 0;
  std::string name = payload_type->Attr(QN_PHONE_PAYLOADTYPE_NAME);
  desc->AddCodec(MediaEngine::Codec(id, name, 0));

Create a Session Wrapper

In libjingle, the Session object handles the common mechanics of the session negotiation (sending and receiving invitations, creating a TransportChannel to manage the data channel, and ending the session), but you must create a higher level wrapper object to manage both of these objects appropriately for a session type. For example, in the file share code, FileShareSession creates and hosts the objects that make and serve the HTTP request, and sends the requests for each file in turn.

The following table describes the most important signals and methods that your controlling classes will need to know about the Session object.

Session Signal or Method Description
Initiate Sends a connection request to another computer.
Accept Accepts an incoming request. The incoming session will be sent to SessionClient::OnSessionCreate.
Reject Rejects an incoming request. The incoming session will be sent to SessionClient::OnSessionCreate.
Terminate Ends a current session.
CreateChannel Creates a TransportChannel object, used to send and receive data from the other party. See "Using the P2PSocket Object" below.
SignalState Sent when the session state changes, as described in the voice chat description.
SignalError Sent when an error occurs.
SignalInfoMessage Sent when a Session object receives an information message specific to the session type (for example, "share complete" when all files have been requested in the file sharing example).
SignalChannelGone Called when the connection has been broken for some reason.
SendInfoMessage Sends an XMPP message to the other party, containing information specific to the session type (for example, file not found in the file sharing application).

Using the TransportChannel Object

Once you have a Session object managing the connection, you can stream data between computers. The top level object managing the data channel is TransportChannel (or a derived class). All data is sent or received across the network using this socket, which manages connectivity and performance for you. libjingle supports two kinds of TransportChannel objects: TCP-type sockets, which provide a lossless data connection with TCP-like functionality that supports redundancy and robustness for data-critical applications like file transfer; and RTP-type sockets, which provide a higher transmission speed for real-time applications like voice chat. How you read and write to the socket depends on what kind of socket you create.

RTP-type Channels
  1. Create the TransportChannel using the Session object. If this is an incoming connection, it is the Session passed to your SessionClient::OnSessionCreate implementation. If this is an outgoing connection, it is the Session object you explicitly created. Create a TransportChannel by calling Session::CreateChannel with "rtp" as the name.
  2. Connect to receive messages from the new socket. You should connect to Socket::SignalWritableState and Socket::SignalReadPacket to receive state messages and received data. See the reference documentation for more information about those methods.
  3. Send data to the other computer by calling TransportChannel::SendPacket; read data that is sent through the TransportChannel::SignalReadPacket signal. See voicechannel.cc for example code.
TCP-type Channels

TCP sockets require several more helper classes and layers of indirection in order to support data redundancy and a stream-like interface. The file share example uses PseudoTcpChannel to expose stream read/write functionality. See File Share Application and StreamInterface for more information.

Supporting Additional Media Engines

libjingle ships with two media engines, Linphone and GIPS VoiceEngine Lite, that enable live audio to be captured and rendered on your computer. If you want to add a new media engine to support audio capturing or rendering, or if you want to support a new media type such as video, you must override the MediaEngine class and the MediaChannel class as described next. (Supporting new media types requires a much more extensive redesign of the supporting classes, as described in Supporting Additional Session Types above.)


The MediaEngine base class represents the components that manage rendering and capturing hardware. Specialized classes extend MediaEngine to support different media hardware. The examples included in libjingle use two MediaEngine specializations called LinphoneMediaEngine, which uses Linphone code, and GipsLiteMediaEngine which uses GIPS VoiceEngine Lite. The MediaEngine class is responsible for handling global audio settings and creating MediaChannel objects.

A customized MediaEngine implementation must include the following:

  • The ability to capture or render media data as required from the computer hardware.
  • The ability to encode and decode media using suitable codecs. Currently, to work with Google Talk, you must support at least one of the following codecs: PCMA, PCMU, G.723, iLBC, ISAC, IPCMWB, EG711U, or EG711A.
  • The ability to understand Real-Time Protocol (RTP). MediaEngine must work with the P2PTransportChannel class, meaning it should be capable of receiving incoming packets by a function call and sending outgoing packets by a callback.
  • The ability to create and manage MediaChannel objects (described below). A MediaChannel object represents an individual media stream. As with MediaEngine, subclass the MediaChannel class to support your specific media components.

The MediaEngine class exposes eight methods that you should override. If your implementation handles a non-audio media type, it should add corresponding methods to handle that medium.

MediaEngine Method Description
CreateChannel Creates a new MediaChannel object, representing a media stream associated with a session. There is one MediaChannel associated with each Session.
GetInputLevel Returns the current input level, used to show your outgoing volume in the UI. This isn't used by libjingle, but can be used by your application.
Init Initializes your media library and populates the vector of codecs that your engine supports (see below).
SetAudioOptions Sets audio options found in the MediaEngineOptions enum. Currently, only one option exists: AUTO_GAIN_CONTROL.

This isn't used by libjingle, but can be used by your application.
SetSoundDevices Sets the sound hardware to use. Your application and your MediaEngine extension should both understand what the integer arguments mean. This isn't used by libjingle, but can be used by your application.
Terminate Cleans up the resources used by the MediaEngine.
codecs Returns a list of codecs supported by the engine.
FindCodec Indicates whether a specific codec is supported by the engine.

Overriding MediaEngine::Init

In your MediaEngine::Init implementation, you should populate the internal codecs_ member vector with Codec structs representing the codecs that your MediaEngine class supports. PhoneSessionClient sorts these codecs in order of preference before sending the session description. The less-than operator used to sort codecs is also encapsulated within the struct; the higher the number, the greater the preference. Create Codec objects with their constructor and push them onto the codecs_ vector as shown here:

  codecs_.push_back(Codec(110, "speex", 8));
  codecs_.push_back(Codec(102, "iLBC", 4));
  codecs_.push_back(Codec(0, "PCMU", 2));

When the PhoneSessionClient class specifies which codec to use, it does so only by the codec's name property . The codec used varies from voice session to voice session.

Customizing the MediaChannel Class

The MediaChannel class represents the data flow between the P2PTransportChannel and the MediaEngine in one-to-one audio conversation, with its own codec and RTP streams. It contains a number of overridable methods for negotiating which codec to use, whether to play or mute audio, and the current audio quality and volume of the voice session.

The MediaChannel class is responsible for sending and receiving data from the media renderers to the peer-to-peer session. When the session component of libjingle receives a packet from the P2PTransportChannel object, it calls MediaChannel::OnPacketReceived. When the MediaChannel object needs to send a packet, it sends it via the NetworkInterface class.

The NetworkInterface class has a single method, SendPacket, which MediaChannel uses to send packets to the P2PTransportChannel object. Other classes can subclass from NetworkInterface and then set themselves as the network interface with MediaChannel::SetInterface.

The overridable functions of the MediaChannel class are:

MediaChannel Method Description
GetCurrentQuality Returns the audio quality. This isn't used by libjingle, but may be used by your application.
GetOutputLevel Returns the volume of the incoming audio on this stream.

This isn't used by libjingle, but by the application.
OnPacketReceived Called when a packet has been read from the network. The MediaChannel should handle the incoming RTP packet and play the audio.
SetCodec Called when a codec for this session has been negotiated. It is passed the codec as a string.
SetPlayout Sets whether or not to play audio from this session.
SetSend Sets whether to send audio over the network from this session. This is used by the Mute feature.