SIP as RTMP - StreamNotFound error

F

fromin

New Member
Hi,

so I am now successfully creating a call to Twilio SIP, the RESTCall/call endpoint returns 200 OK as well as the getStatus, which is returning 200 OK with body "ESTABLISHED" during the whole call that lasts about a minute (I have a dumb one two three app there saying that all over again for now) and then it starts returning NO SUCH CALL when it ends, so it looks like it's listening correctly.

However, the RTMP player says "StreamNotFound" when I try to play it during the call.

What could be wrong?

Thanks,
Martin
 
Max

Max

Administrator
Staff member
Try these settings in flashphoner.properties:
Code: 
video_enabled=false
generate_av_for_ua=mySipGateway
Here mySipGateway is S section of SDP returned by Twilio in the 200 OK response.
Example:
Code: 
o=...
s=mySipGateway
Try to set these settings and restart WCS.
If it does not help, please send us logs, dumps and actual flashphoner.properties config.
 
Max

Max

Administrator
Staff member
Hello
Try to set
Code: 
generate_av_for_ua=Twilio Media Gateway
in flashphoner.properties
This setting requires WCS restart
Code: 
service webcallserver restart
If you have any restart issues, please read the troubleshooting guide.

If it does not help, please create a pcap dump.
We will check.
 
F

fromin

New Member
It didn't help. Can you give me exact tcpdump command (with ports etc.) and tell me what command(s) I need to run to fill the pcap file (last time I handed it to you you said it was empty)
 
F

fromin

New Member
It doesn't (your call ends immediately, mine at least holds for the length it should so it is receiving the signal somehow). I am sending the credentials to logs@ with additional info
 
Max

Max

Administrator
Staff member
Ok. We will test this on your server and inform you about results. Please keep ssh access open.
 
Max

Max

Administrator
Staff member
We have checked and configured your server. Please check.

Was
1. Local WCS's RTMP was not working on port 1935
Code: 
flashphoner.log: ERROR Server - main Error while init RTMP! Failed to bind to: 0.0.0.0/0.0.0.0:1935
2. In flashphoner.properties, ulaw codec only was set. It is root cause why stream was not published.

Now
1. RTMP порт was changed to 1936 in server.properties and publishing to rtmp://localhost:1936/live works.
2. SIP as RTMP works with Twilio with default settings +
Code: 
generate_av_for_ua=Twilio Media Gateway
in the flashphoner.properties:
Code: 
port_from=30000
port_to=31000
media_port_from=31001
media_port_to=32000
waiting_answer=60
user_agent=Flashphoner/1.0
balance_header=balance
cost_header=cost
video_enabled=true
generate_av_for_ua=Twilio Media Gateway
domain=
outbound_proxy=
outbound_port=
log_level=5
enable_context_logs=false
rtp_activity_detecting =true,60
sip_msg_listener=com.flashphoner.sdk.sip.ChangeCallIdListener
call_record_listener=com.flashphoner.server.client.DefaultCallRecordListener
dtmf=rfc2833
auto_login_url=/usr/local/FlashphonerWebCallServer/conf/account.xml
get_callee_url=/usr/local/FlashphonerWebCallServer/conf/callee.xml
codecs=opus,alaw,ulaw,g729,speex16,g722,mpeg4-generic,telephone-event,h264,vp8,flv,mpv
#codecs=ulaw
codecs_exclude_sip=mpeg4-generic,flv,mpv
codecs_exclude_streaming=flv,telephone-event
codecs_exclude_sip_rtmp=opus,g729,g722,mpeg4-generic,vp8,mpv
on_record_hook_script=on_record_hook.sh
rtmp_transponder_stream_name_prefix=rtmp_
Note that by default
Code: 
rtmp_transponder_stream_name_prefix =rtmp_
For example, if you set RTMP stream name to stream1 it will be prefixed with rtmp_ like:
Code: 
rtmp://localhost:1936/live/rtmp_stream1
 
Last edited:
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