I have a Grandstream GDS3710 video doorbell and a GS3510 video intercom connected to FreePBX through pjsip, and when the GDS3710 calls directly the GS3510 to it’s IP there is a preview of the video on the screen before answering but through FreePBX it is not working.
I tried the solution by @danielf below but unfortunately without success:
I’m getting the added header in the logs, but I don’t know if it is because the URL of the streams seems to have changed, there’s only an RTSP stream I tried with it but without success.
I’ve seen there may be something to configure with pjsip to allow the preview video but I didn’t find exactly what to do.
I’ve tried a couple of things without success, is there any way to see what could be missing during the call to make this work?
You do not need to see anything when you access this page directly. Probably your settings of the GDS3710 not working right. What is the other device that you are using to display the snapshots? the GS3510 is a SIP intercom without a screen to display the snapshots.
Can you elaborate more on your phone network topology?
Oops, again I did a mistake, the device I’m using is a GSC3570
The small intercom screen.
I wrote the wrong reference as I also have a GSC3510 but not working with this.
At this time I only have the GDS3710 and GSC3570 together through FreePBX, I plan to add more devices and make a ring group later, that’s why I’m trying to fix this before going into a more complex setup.
I’ve been in touch with Grandstream support since the last message, for them the preview snapshot should work on this device.
They asked me to send them some logs of the system and for them the problem is that:
there is no RTP traffic, the server doesn’t send RTP.
They asked me to try with a STUN server configured but no success.
Is there something that may need an adjustment (or may be misconfigured) in FreePBX?
Most of the problems in early media are that the Asterisk (not the Freepbx) does not support it well or not at all. As far as I know, the PJSIP is not supporting early media video (I am sure that @jcolp would be able to clarify more on this topic).
If you will connect the devices to another SIP proxy or even another pbx it would work well with early media video. Maybe it would be better to connect the devices to a Grandstream pbx and then connect it through a SIP trunk to the Asterisk (it it’s needed).
Your best solution in the current situation is to send the call of the GDS3710 directly to the GSC3570 and not through the Asterisk.
PJSIP doesn’t care either way really, someone added support previously to the core to support it (as it didn’t) but that may or may not work. It’s not common usage, and would require someone to trace and understand the complete flow of media through Asterisk to investigate.