SIP as RTMP not working again

[fromin]

Hi,

so we tried to duplicate what we had working on our dev server to our live, but the call doesn't seem to be established correctly (after the first getStatus, it already returns "NO SUCH CALL" although the call should take over 100 seconds). I am sending an e-mail to logs@ now as well.

Thanks
Martin

 

[Max]

Hello
Your configuration looks correct.
According your logs, your RTMP server does not receive RTMP stream.
We have checked your RTMP server with FMLE. As you can see, connection was failed.
Please check your RTMP endpoint and make sure it is working.

 

[fromin]

If it's the problem just with RTMP server not receiving the stream, why does the getStatus say "NO SUCH CALL" after 2 seconds of the call failing although the SIP call should take 100 seconds?

 

[Max]

Because SIP call is linked with RTMP stream. If RTMP stream is failed, associated SIP call will be terminated immediately.
If your SIP call is alive after RTMP fail, it looks like a bug because SIP call should be terminated.
However in logs I see that call is terminated with BYE request.

 

[fromin]

Ok I fixed the RTMP and it works now. Only thing is that getStatus now returns the stream ID, but on the other server it returns "ESTABLISHED" - is that because I have a newer version on this server?

 

[Max]

Yes it looks like be a bug with new version of server. We will check.

 

[fromin]

Ok so I made some adjustments, the RTMP looks like it's capturing it correctly, but I cannot hear any sound, although the call is exactly 1 minute 16 seconds which it should be. Please try - I set up a dummy call that repeats "One two three" for 76 seconds.

 

[Max]

We have tested this with your server. It works fine.
We tested on default samples:
1. SIP as RTMP
/demo2/sip-as-rtmp

2. Flash Streaming
/demo2/flash-streaming

As you can see, WCS adds prefix rtmp_ for stream name stream123. And you have to play stream rtmp_stream123.
You can disable this prefix in flashphoner.properties.

 

[fromin]

Hmm, for some reason if I use the rtmp://dev.***.com as you did, it works, but with the other RTMP server (rtmp://staging.***.com), I'm not receiving the sound as I described earlier.
I need this to work with staging

 

[fromin]

The RTMP server is Wowza and I just checked that the Wowza configuration is identical

 

[Max]

WCS sends audio in AAC codecs in latest builds.
Perhaps if you deploy the same Wowza version on your stage host, it would work properly.
Other options:
1. Switch to old engine in flashphoner.properties:

sip_as_rtmp_use_new_engine=false

In such case WCS will use G.711 codec received from Twilio
2. Use speex codec

force_rtmp_audio_codec=speex16

You can try these two options in combination.
These settings require WCS restart.

 

[fromin]

So I tried all of the combinations (restarted after modifying of course); results:

sip_as_rtmp_use_new_engine=false
****
streamnotfound when I tried to play the stream

force_rtmp_audio_codec=speex16
*****
no sound (therefore no change when compared to the original settings)

force_rtmp_audio_codec=speex16
sip_as_rtmp_use_new_engine=false
*****
streamnotfound when I tried to play the stream

 

[fromin]

And yes I'm using the rtmp_ prefix when trying to play it (I'm trying it via the WCS dashboard)

 

[fromin]

Any news? I need to get this done ASAP

 

[Max]

You can update to previous build
2101
And check if it works for you with:

sip_as_rtmp_use_new_engine=false

or

sip_as_rtmp_use_new_engine=true

The issue is under investigation. I will inform you if we have any news.

 

[Max]

We have tested with latest build 2172 and our Twilio account.
1. It works fine with default settings.
2. It works with

sip_as_rtmp_use_new_engine=false

Perhaps you have an issue with your Wowza configuration. Please test the build.