...
- Video: H.264, VP8
- Audio: G.711, Speex, Opus
Supported RTMP codecs
- Video: H.264
- Audio: AAC, G.711, Speex
...
REST-methods and response statuses
REST-method | Example of REST-query | Example of REST-response body | Response status | ||||||||||||||
---|---|---|---|---|---|---|---|---|---|---|---|---|---|---|---|---|---|
/call/startup |
| {} | 200 - The call is accepted for processing 409 - Conflict with an existing RTMP URL | ||||||||||||||
/call/find |
|
| 200 - call is found | ||||||||||||||
/call/find_all | {} |
| 200 - calls are found | ||||||||||||||
/call/terminate |
| 200 - call is terminated | |||||||||||||||
/call/send_dtmf |
| 200 - DTMF is sent |
404 - call is not found
language | js |
---|---|
theme | RDark |
404 - call is not found |
Parameters
Parameter name | Description | Example |
---|---|---|
callId | SIP Call ID - a unique identifier string | Xq2tlLcX89tTjaji |
callee | SIP callee | 10001 |
toStream | Name of the stream on the WCS server the call is published to | call_stream1 |
rtmpUrl | RTMP URL - address of the RTMP server | rtmp://rtmp-server.flashphoner.com:1935/live Here live - is the name of the RTMP application. |
rtmpStream | Name of the RTMP stream on the RTMP server | streamName2 |
hasAudio | If true, SDP will have the 'sendrecv' parameter in audio. If false, it gets 'recvonly'. | true |
hasVideo | If true, SDP will have the 'sendrecv' parameter in video. If false, it gets 'recvonly'. | true |
status | Call status on the WCS server | ESTABLISHED The complete list of statuses is available in the Call Flow (see the CallStatusEvent method). |
sipStatus | Associated SIP-status | 200 |
rtmpStreamStatus | Status of the RTMP stream | RTMP_STREAM_ACTIVE RTMP_STREAM_WAIT - RTMP-stream is initializing |
caller | SIP caller | |
visibleName | Displayed name of the caller | |
mediaProvider | NOT USED | NOT USED |
SDP parameters recvonly and sendrecv
...
Code Block | ||||
---|---|---|---|---|
| ||||
session.createStream({constraints:{audio:true,video:false}}).play(); |
3. It's impossible to make a SIP call if 'SIP Login' and 'SIP Authentification name' fields are incorrect
...
Solution: according to the standard, 'SIP Login' and 'SIP Authentification Authentication name' should not contain any of unescaped spaces and special symbols and should not be enclosed in angle brackets '<>'.
...
Code Block | ||||
---|---|---|---|---|
| ||||
sipLogin='Ralf_C12441' sipAuthenticationName='Ralf_C' sipPassword='demo' sipVisibleName='Ralf C' |
4. RTP traffic buffering should be enabled in some cases when republishing SIP as Stream or SIP as RTMP
Symptoms: audio and video may be out of sync when playing a SIP call stream
Solution: update WCS to build 5.2.1910 and enable RTP traffic buffering
Code Block | ||
---|---|---|
| ||
rtp_in_buffer=true |