SIP as RTMP: Call is estableshed but RTMP content is not published

NickEm

New Member
I have configured two SIP accounts. One account I use in WCS participant, second account I use in X-Lite. Also I started and added RMTP receiver details to RTMP target Details.
I set up call between these accounts (WCS is initiator).
In the status line I see "established", from the X-Lite I'm able to share audio and video but I don't see any results in the RTMP player and in my receiver.
I checked ports that are used in flashphoner.properties, all of them are opened.
I checked also IP adrresses in the same properties. Everithing look fine. Also in managers logs I see RTMP_STREAM_ACTIVE wich gives me some confidence that RTMP stream configuration is OK.
I notice that you seems to use h264 video codec while from X-Lite perspective I'm producing h263, could this be an issue?
If it so, what about just audio, I use codecs G711 uLaw, G.722.
What else could be incorrect?
Also properties in request "hasAudio", "hasVideo" do they really matter in case that we want just listen for content from WCS perspective?
FYI: I'm using 5.0.2099 version of WCS
 
Last edited:

Max

Administrator
Staff member
I'm producing h263, could this be an issue?
Yes. H.263 is not currently supported for this SIP as RTMP case.
You can try to turn off video codecs in your Xlite softphone or disable video on WCS
flashphoner.properties
Code:
video_enabled=false
 

Max

Administrator
Staff member
What else could be incorrect?
If you are using Amazon EC2 instance and external SIP gateway, you have to add this setting in flashphoner.properties
Code:
client_mode=false
This setting inserts external Amazon EC2 address into the SIP SDP.
 

Max

Administrator
Staff member
Some SIP gateways require to send audio / video packets.
WCS can send silence audio and static video using internal RTP generator.
You have to specify SDP s= value in the setting in flashphoner.properties
Code:
generate_av_for_ua=mySipGateway
If you share your SDP from flashphoner.log, I will able to advice how this setting should be set.
 

NickEm

New Member
Yes. H.263 is not currently supported for this SIP as RTMP case.
You can try to turn off video codecs in your Xlite softphone or disable video on WCS
flashphoner.properties
Code:
video_enabled=false
Tried to do from WCS side, but same result. This didn't help
 

NickEm

New Member
If you are using Amazon EC2 instance and external SIP gateway, you have to add this setting in flashphoner.properties
Code:
client_mode=false
This setting inserts external Amazon EC2 address into the SIP SDP.
Np, I don't use EC2. I use Ubuntu on VM.
 

NickEm

New Member
Some SIP gateways require to send audio / video packets.
WCS can send silence audio and static video using internal RTP generator.
You have to specify SDP s= value in the setting in flashphoner.properties
Code:
generate_av_for_ua=mySipGateway
If you share your SDP from flashphoner.log, I will able to advice how this setting should be set.
If you are talking about something like CallConnection object in flashphoner.log than sdp field in it always null.
 

Max

Administrator
Staff member
Please change this setting in WCS_HOME/bin/setenv.sh
Code:
-Dnode.enable_stdout=false
to
Code:
-Dnode.enable_stdout=true
then restart WCS server
Code:
service webcallserver restart
then make this test again and send us zipped logs and configs to logs@flashphoner.com. We will check.
Code:
WCS_HOME/logs/server_logs/flashphoner.log
WCS_HOME/logs/flashphoner_manager.log
WCS_HOME/logs/server-stdout.log
WCS_HOME/conf
log.pcap
 

NickEm

New Member
Please change this setting in WCS_HOME/bin/setenv.sh
Code:
-Dnode.enable_stdout=false
to
Code:
-Dnode.enable_stdout=true
then restart WCS server
Code:
service webcallserver restart
then make this test again and send us zipped logs and configs to logs@flashphoner.com. We will check.
Code:
WCS_HOME/logs/server_logs/flashphoner.log
WCS_HOME/logs/flashphoner_manager.log
WCS_HOME/logs/server-stdout.log
WCS_HOME/conf
log.pcap
Thanks for support, I have sent you logs
 

Max

Administrator
Staff member
We have checked your logs.
There is no RTP traffic in your log.pcap.
Please try the following settings in flashphoner.properties:
1. Generate traffic for your SIP gateway / User Agent. It may help to path through the NAT and receive RTP packets.
Code:
generate_av_for_ua=X-Lite release 4.9.7.1 stamp 83372
2. The same in combination with client_mode=false
Code:
generate_av_for_ua=X-Lite release 4.9.7.1 stamp 83372
Code:
client_mode=false
These settings require WCS restart.
 
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