Very high level API for constructing SIP UA applications. More...
Modules | |
| PJSUA-API Basic API | |
| Basic application creation/initialization, logging configuration, etc. | |
| PJSUA-API Signaling Transport | |
| API for managing SIP transports. | |
| PJSUA-API Accounts Management | |
| PJSUA Accounts management. | |
| PJSUA-API Calls Management | |
| Call manipulation. | |
| PJSUA-API Buddy, Presence, and Instant Messaging | |
| Buddy management, buddy's presence, and instant messaging. | |
| PJSUA-API Media Manipulation | |
| Media manipulation. | |
| PJSUA-API Video | |
| Video support. | |
Detailed Description
PJSUA API is very high level API for constructing SIP multimedia user agent applications. It wraps together the signaling and media functionalities into an easy to use call API, provides account management, buddy management, presence, instant messaging, along with multimedia features such as conferencing, file streaming, local playback, voice recording, and so on.
C/C++ Binding
Application must link with pjsua-lib to use this API. In addition, this library depends on the following libraries:
Codecs: null, PCMU, PCMA, G722, G7221, G726-32, opus, G729, iLBC, L16, L16-256: Enables or disables codecs and sets the codec priority. To disable, prefix a codec name with an '!' Earlier entries have higher priority. Not all codecs apply to all models of phones. Openvpnenable: 0, 1.
- After the initial announcement played, Asterisk went back to sending audio in codec G711 and I then could here the echo test. Other notes: - If on Asterisk I make G722 the first option than everything works fine.; - With exactly the same codec priority as set on the T46g, my Bria Android client worked just fine and I could hear the announcement.
- I am trying to change the priority of codecs used in pjsip android. I am able to get the codecs priority but after changing the codecs priority,it is not reflecting back.
- How we can compile pjsip-2.5.5 with opus codec for android multiple architectures 0 I want to develop an android VOIP app using PJSIP stack by its high level api i.e.
- pjsip-ua,
- pjsip-simple,
- pjsip-core,
- pjmedia,
- pjmedia-codec,
- pjlib-util, and
- pjlib,
so application must also link with these libraries as well. For more information, please refer to Getting Started with PJSIP page.
Few samples are provided:
- page_pjsip_sample_simple_pjsuaua_c
Very simple SIP User Agent with registration, call, and media, using PJSUA-API, all in under 200 lines of code. - page_pjsip_samples_pjsua
This is the reference implementation for PJSIP and PJMEDIA. PJSUA is a console based application, designed to be simple enough to be readble, but powerful enough to demonstrate all features available in PJSIP and PJMEDIA.
Please refer to PJSUA-API Basic API on how to create and initialize the API. And then see the Modules on the bottom of this page for more information about specific subject.
Skip to end of metadataGo to start of metadataAsterisk supports a variety of audio and video media. Asterisk provides CODEC modules to facilitate encoding and decoding of audio streams. Additionally file format modules are provided to handle writing to and reading from the file-system.
The tables on this page describe what capabilities Asterisk supports and specific details for each format.
- Enabling specific media support
- Channel driver configuration
- Audio Support
There are three basic requirements for making use of specific audio or video media with Asterisk.
- The Asterisk core must support the format or a module may be required to add the support.
- Asterisk configuration must be modified appropriately.
- The devices interfacing with Asterisk must support the format and be configured to use it.
Module compilation and loading
For audio or video capabilities that require a module - you should make sure that the module is built and installed on the system.
See the section on Using Menuselect to Select Asterisk Options if you need help figuring out how to get a module built and then section on Configuring the Asterisk Module Loader to verify that a module gets loaded when Asterisk starts up.
Channel driver configuration
Audio or video capabilities for Asterisk are used on a per channel or per feature basis. To tell Asterisk what CODECs or formats to use in a particular scenario you may need to configure your channel driver, or modify configuration for the feature itself.
We'll provide two examples, but you should look at the documentation for the channel driver or feature to better understand how to configure media in that context.
Configuring allowed media for a PJSIP endpoint
We set the option 'allow' to a string of values '!all,ulaw'.
- The value '!all' means 'Disallow all' and is identical to 'disallow=all'. This tells Asterisk to disallow all codecs except what we further define in the allow option.
- The value 'ulaw' instructs Asterisk to allow ulaw audio during media negotiation for this endpoint.
See the section Configuring res_pjsip for more information on the PJSIP channel driver.
Configuring app_voicemail file formats for recordings
In the general section of voicemail.conf you can set the formats used when writing voicemail to the file-system. We set the option 'format' to a string of file format names.
- The value 'wav49' represents GSM in a WAV|wav49 container.
- The value 'wav' represents SLIN in a wav container.
- The value 'gsm' represents GSM in straight gsm format.
Endpoint device configuration
Configuring your particular device is outside the scope of the Asterisk documentation.
Consult your devices user/admin manual to find out where you define codecs or media to be used.
For VoIP desk phones there are typically two places to look for media configuration.
- The web GUI for the phone.
- The provisioning files that are pulled down by the phones on your network.
A variety of audio capabilities are supported by Asterisk.
| Name | Config Value | Capability: (P)assthrough | CODEC Module | Format Module | Distributed w/ Asterisk? | Commercial License Required? |
|---|---|---|---|---|---|---|
| ADPCM | adpcm | T | codec_adpcm | format_vox | YES | NO |
| G.711 A-law | alaw | T | codec_alaw | format_pcm | YES | NO |
| G.711 ยต-law | ulaw | T | codec_ulaw | format_pcm | YES | NO |
| G.719 | g719 | P | n/a | format_g719 | YES | NO |
| G.722 | g722 | T | codec_g722 | format_pcm | YES | NO |
| G.722.1 Siren7 | siren7 | T | codec_siren7 | format_siren7 | Codec(NO) Format(YES) | NO |
| G.722.1C Siren14 | siren14 | T | codec_siren14 | format_siren14 | Codec(NO) Format(YES) | NO |
| G.723.1 | g723 | T | codec_g723 | format_g723 | Codec(NO) Format(YES) | YES(hardware required) |
| G.726 | g726 | T | codec_g726 | format_g726 | YES | NO |
| G.726 AAL2 | g726aal2 | T | codec_g726 | format_g726 | YES | NO |
| G.729A | g729 | T | codec_g729a | format_g729 | Codec(NO) Format(YES) | YES |
| GSM | gsm | T | codec_gsm | format_gsm | YES | NO |
| ILBC | ilbc | T | codec_ilbc | format_ilbc | YES | NO |
| LPC-10 | lpc10 | T | codec_lpc10 | n/a | YES | NO |
| SILK | silk | T | codec_silk | n/a | Codec(NO) Format(YES) | NO |
| Speex | speex | T | codec_speex | n/a | YES | NO |
| Signed Linear PCM | slin | T | codec_resample | format_sln | YES | NO |
| Ogg Vorbis | n/a | n/a | n/a | format_ogg_vorbis | Codec(NO) Format(YES) | NO |
| Opus | opus | T | codec_opus | n/a | Codec(NO) Format(YES) | NO |
| wav (SLIN) | wav | T | n/a | format_wav | YES | NO |
| WAV (GSM) | wav49 | T | n/a | format_wav_gsm | YES | NO |
Speex Support
Asterisk supports 8, 16, and 32kHz Speex. Use of the 32kHz Speex mode is, like the other modes, controlled in the respective channel driver's configuration file, e.g. chan_sip's sip.conf or PJSIP's pjsip.conf.
Download Download Free Pjsip Codec Priority Mac
Signed Linear PCM
Asterisk can resample between several different sampling rates and can read/write raw 16-bit signed linear audio files from/to disk. The complete list of supported sampling rates and file format is found in the expansion link below:
Asterisk File format | |
|---|---|
8kHz | .sln |
12kHz | .sln12 |
16kHz | .sln16 |
24kHz | .sln24 |
32kHz | .sln32 |
44.1kHz | .sln44 |
48kHz | .sln48 |
96kHz | .sln96 |
192kHz | .sln192 |
Users can create 16-bit Signed Linear files of varying sampling rates from WAV files using the sox command-line audio utility.
In this example, an input WAV file has been converted to Signed Linear at a depth of 16-bits and at a rate of 32kHz. The resulting output.sln file is then renamed output.sln32 so that it can be processed correctly by Asterisk.
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You'll notice the CODEC module column is missing. Video transcoding or image transcoding is not currently supported.
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| Name | Config Value | Capability: (P)assthrough | Format Module | Distributed w/ Asterisk |
|---|---|---|---|---|
| JPEG | jpeg | P | format_jpeg | YES |
| H.261 | h261 | P | n/a | YES |
| H.263 | h263 | P | format_h263 | YES |
| H.263+ | h263p | P | format_h263 | YES |
| H.264 | h264 | P | format_h264 | YES |
| VP8 | vp8 | P | n/a | YES |
| VP9 | vp9 | P | n/a | YES |