Trouble connecting SIP to RTMP with Twilio, Wowza and Flashphoner

[Sam Udotong]

I'm having trouble connecting to SIP through WCS 5 -- even in the :9091/dashboard.xhtml?demo2=demo2/sip-as-rtmp demo page, I get this message

AOOsuUF-zcH3Fp3-eKTbtGW-Q8xGNeb >>> rtmp://host:1935/live
FINISHED

but when connecting through the RTMP player, I get StreamNotFound.

I've followed these guides/posts closely and have not been able to get a fix:

https://www.codeproject.com/Articles/1058144/Taking-audio-stream-from-Twilio-by-SIP-and-sending
https://flashphoner.com/docs/wcs5/w...-guide/index.html?sip_as_rtmp_with_twilio.htm
http://forum.flashphoner.com/threads/sip-as-rtmp-streamnotfound-error.10805/

I seem to be getting getting an INTERNAL_SIP_ERROR when executing the JSON POST request to the WCS server to start the audio stream. I've confirmed that I am able to get the SIP audio stream if I call into my twilio number with a real cellphone, and also through a softphone (Zoiper). However when I use the correct parameters in JSON POST request and then try to connect, it leaves me with StreamNotFound.

I've sent my conf and logs over to the logs@ email address- any times?

 

[Max]

Hello
From your logs we are able to see

X-Twilio-Error: 32201 Authentication failure - source IP Address not in ACL.

You have to add WCS external IP address into ACL (Twilio white list).

 

[Sam Udotong]

Thanks Max. I added the IP address into the Twilio white list which made a real difference - Now from the sip-as-rtmp demo page the call is established for around 20 seconds, which is the identical behavior to what happens when I call in from the softphone.

However I'm still not able to connect via the RTMP Player - I'm getting the StreamNotFound error. Any thoughts?

 

[Max]

Please share your flashphoner.log file or send this file to logs@flashphoner.com
Note, this log file is rotated hourly. Make sure it has logs of the latest test.
Are you trying to play stream name rtmp_stream1? Because WCS server adds rtmp_ prefix by default.
With latest builds
1. Try to revert codec settings to defaults.
codecs=opus,alaw,ulaw,g729,speex16,g722,mpeg4-generic,telephone-event,h264,vp8,flv,mpv
codecs_exclude_sip=mpeg4-generic,flv,mpv
codecs_exclude_streaming=flv,telephone-event
codecs_exclude_sip_rtmp=opus,g729,g722,mpeg4-generic,vp8,mpv
2. Try to find and set

video_enabled=false

in flashphoner.properties then restart WCS server
With previous build
If it does not help, you can try one of previous build 2101 as recommended here:
http://forum.flashphoner.com/threads/sip-as-rtmp-broadcasting-problem.10884/#post-12363

 

[Sam Udotong]

My settings where already reverted as is the case in point #1. However I switched video_enabled from true to false as per point #2. Also I kept this setting:

generate_av_for_ua = Twilio Media Gateway

I restarted WCS server and retested in the sip-as-rtmp demo page (I am playing stream1 and letting WCS add the rtmp_ prefix on its own).

That fixed the issue! Thanks so much.