No video when call is routed through FreePBX

Hello guys,

I am trying to integrate a SIP endpoint that is not a standard phone but rather an interface that speaks with the door “bell”, purpose being (obviously) to see and speak with people at the front door.

The device (it’s a “Ritto Wiser Door Gateway”) allows to create (exactly) one SIP account for a phone to connect to it. If I configure my phone (a Grandstream GXV 3380) to connect to it directly, I get both audio and video.

To actually make use of the setup, I want to integrate this device with FreePBX. I need FPBX to connect to the account created on the device so I used a trunk setup.

My idea was now to have an inbound route to a ring group to be able to let someone in from multiple phones. Works, with audio only though. Same thing, if I use an inbound route that directly goes to only one extension.

From what I understand now, video is not allowed for calls coming in from a trunk (i.e. the “from-pstn” context). I used the config editor to add the following to pjsip_custom_post.conf:

[ritto-gw](+)
allow=h264,h263,h261,h263p

(ritto-gw is, of course, the name of the trunk for the device)

This didn’t help. Video is also enabled in the general SIP settings: Video calls between the phones internally work perfectly fine.

The console tells me that video is allowed for the endpoint:

allow                              : (alaw|ulaw|h264|h263|h261|h263p)

SIP looks like this from the door device to FPBX:

v=0
o=- 3791703424 3791703427 IN IP4 172.30.255.8
s=pjmedia
b=AS:352
t=0 0
a=X-nat:0
m=audio 4140 RTP/AVP 8 96
c=IN IP4 172.30.255.8
b=TIAS:64000
a=rtcp:4141 IN IP4 172.30.255.8
a=rtpmap:8 PCMA/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-16
a=sendrecv
m=video 4142 RTP/AVP 97
c=IN IP4 172.30.255.8
b=TIAS:256000
a=rtcp:4143 IN IP4 172.30.255.8
a=rtpmap:97 H264/90000
a=fmtp:97 profile-level-id=42e01e; packetization-mode=1
a=sendrecven

And then from FPBX to the phone like this:

v=0
o=- 1134391926 1134391926 IN IP4 172.30.1.9
s=Asterisk
c=IN IP4 172.30.1.9
t=0 0
m=audio 16374 RTP/AVP 8 0 9 111 3 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:9 G722/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

I did this with sngrep. If I interpret this correctly, the video gets “lost” on the way to my actual phone :expressionless:

I also have a SIP log taken from a packet capture when the phone is directly connected to the door device:

https://pastebin.com/58nxMQtY

Sorry for the long post, that’s all I have and did to find the problem.

Full disclosure also: I already posted this a while ago as part of a post where I mixed a lot of different issues together. This is hopefully “untangles” this a bit. Thanks.

I don’t know of anyone that’s ever gotten video calls to a ring group to work. if it’s going to, an RG is the way to go, but I don’t know that I’ve ever heard of anyone even asking.

From the vast amount of lurking I’ve done, I do remember that we’ve had problems in the past where a connection from the PBX to a phone will settle on the wrong codec and not move away. Something simple you can try is changing the order of the codecs so that the H2XX codecs are “preferred”, just in case it settling too quickly.

I actually read a thread about this from a while ago but it has no usable information on the solution. It doesn’t work without a ring group though, so this is probably not the source of the problem.

How can I change the codec preference? I know I can change it in the general SIP settings but separately for audio and video but not e.g. to prefer h264 over alaw.

In your extension setup, there’s a re-orderable list of codecs that are listed in order of preference. You should be able to move alaw below H264 that way.

I can see a codec list for the trunk setup under pjsip settings. That list does not include any video codecs at all.

I can see no codec list for extensions though (I thought this is what the general pjsip settings are for?)

1 Like

This topic was automatically closed 31 days after the last reply. New replies are no longer allowed.