Warning: This sample includes GPLv2 code (PJSIP library). Use of a GPLv2 library results in this sample app being GPLv2 also.; This sample app and its interface are experimental and unsupported.

The streamMedia sample

This sample demonstrates an audio streaming over IP app. It uses:

a 3rd-party library that implements RTP protocol (PJSIP);
the audio services with .adef bindings and related .cdef requires provides, and sources sections.

The streamMedia sample implements an experimental service app exposing the Stream Media API. Moreover, it also implements one additional test app beside the service app: the streamMediaTest app that contains two executables that show how to use the Stream Media API. As this sample provides two apps, one for the service and another one for the test, you can find two .adef files:

streamMedia.adef for the service;
streamMediaTest.adef for the test app.

The Stream Media API allows the user to connect the audio streaming over IP interface to other audio interfaces via audio connectors. Its usage follows the same rules as the other audio interfaces (like Mic, Speaker, MdmTx, ...), please refer to Audio API and Stream Media API documentation.

Go to the sample code.

Run this to install the sample app:

$ cd apps/sample/streamMedia
$ make wp85
$ update streamMedia.wp85.update <ip address>

where ip address is the address of your target device.

If you're not using a Sierra Wireless WP85xx module, replace wp85 with the appropriate identifier for your target.

Note: The streamMedia application has only been tested on WP85 and AR8652.

The streamMediaTest app

The streamMediaTest app contains two executables that show how to use the Stream Media API.

Note that these applications must be terminated using :

app stop streamMediaTest

Otherwise the audio interface will not be disconnected and closed properly.

rtpLoopback

rtpLoopback is a quick demonstration on how to use the Stream Media API. It creates a local RTP session by getting audio from the microphone and sending it through RTP to the speaker.

To run the application, use the following command :

app runProc streamMediaTest --exe=rtpLoopback

rtpTest

rtpTest implements the use case pictured in Stream Media API documentation. There are two test cases : one for the modem side (MODEM_PEER) and the other for the microphone and speaker side (AUDIO_PEER).

To run the application, use the following command :

app runProc streamMediaTest --exe=rtpTest -- <test case> <remote ipv4 addr> <tel number>

For instance, to run an audio peer that will send audio from the microphone to the remote address and play the received RTP packets on the speaker :

app runProc streamMediaTest --exe=rtpTest -- AUDIO_PEER 192.168.10.2

To run the modem peer that sends audio from RTP to a modem call and from this modem call to RTP :

app runProc streamMediaTest --exe=rtpTest -- MODEM_PEER 192.168.2.5 0123456789