Transcoder
Documentation based on repostory git version commit 2.0.2-5-g0c973c23
Module Transcoder
The purpose of the Transcoder module is to transparently transcode from one audio codec to another to make the communication possible between clients that do not share the same set of supported codecs. Concretely, it adds all missing codecs into the INVITEs it receives, and adds codecs matching the original INVITE into the 200Ok. Rtp ports and addresses are masqueraded so that the streams can be processed by the proxy. The transcoding job is done in the background by the Mediastreamer2 library, as consequence the set of supported codecs is exactly the the same as the codec set supported by Mediastreamer2, including the possible plugins you may installed to extend Mediastreamer2.
WARNING: this module can conflict with the MediaRelay module as they are both changing the SDP. Make sure to configure them with different to-domains or from-domains filter if you want to enable both of them.
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Configuration options:
| Name | Description | Default Value | Type |
|---|---|---|---|
| enabled | Indicate whether the module is activated. | false | Boolean |
| filter | A request/response enters module if the boolean filter evaluates to true. Ex: from.uri.domain contains 'sip.linphone.org', from.uri.domain in 'a.org b.org c.org', (to.uri.domain in 'a.org b.org c.org') && (user-agent == 'Linphone v2') | BooleanExpr | |
| jb-nom-size | Nominal size of RTP jitter buffer, in milliseconds. A value of 0 means no jitter buffer (packet processing). | 0 | Integer |
| rc-user-agents | Whitespace separated list of user-agent strings for which audio rate control is performed. | StringList | |
| audio-codecs | Whitespace seprated list of audio codecs, in order of preference. The 'telephone-event' codec is necessary for inband DTMF processing. | speex/8000 amr/8000 iLBC/8000 gsm/8000 pcmu/8000 pcma/8000 telephone-event/8000 | StringList |
| remove-bw-limits | Remove the bandwidth limitations from SDP offers and answers | false | Boolean |
| block-retransmissions | If true, retransmissions of INVITEs will be blocked. The purpose of this option is to limit bandwidth usage and server load on reliable networks. | false | Boolean |