Problems with video calls



  • Asterisk PBX 20.8.1
  • PJSIP Channel


type = transport
protocol = udp
bind =
tos = af31

type = endpoint
transport = tp-udp-internas
context = dp_internas
allow = !all,ulaw,alaw,g722,h264
direct_media = no
trust_id_outbound = yes
device_state_busy_at = 1
dtmf_mode = rfc4733
send_pai = yes
send_rpid = yes
language = es
tos_audio = ef
tos_video = af41

rtp_symmetric = yes
force_rport = yes
rewrite_contact = yes
rtp_timeout = 60
rtp_timeout_hold = 60
rtp_keepalive = 90


auth = 7103-auth
aors = 7103
context = dp_intercom_ops01
callerid = GSC3570 OPS.01 <7103>
mailboxes = 7103@buzones

auth = 7104-auth
aors = 7104
context = dp_intercom_ops01
callerid = GDS3710 OPS.01 <7104>
mailboxes = 7104@buzones

Related to [PJSIP] - Without video but with audio

Remote headquarters equipment:

  • Router (IPsec to PBX)
  • Unifi Dream Machine ( nat ago)
  • Grandstream GDS3710 - SIP Extension 7104 (Intercom)
  • Grandstream GSC3570 - SIP Extension 7103

The PBX is an instance in Google Cloud with Public and Private IP, and the remote offices (IP phones, intercom, etc.) connect via IPsec to the PBX through a direct PBX-router tunnel.
The end devices know the private IP of the PBX without NAT.

All this for the audio is working without problems. But we have a problem with the GSC3570 and GDS3710 that we can’t see a solution to:

  • When someone rings the doorbell of the GDS3710 ext. 7104 a video call is made to GSC3570 ext. 7103, which there are 2 issues:

1- When the GSC3570 has automatic answer configured, the audio and video work correctly.

2- When the GSC3570 (as desired) does NOT have automatic answer configured, the audio works, the video works while it is ringing, but once you pick up the call from the GSC3570 it runs out of video (the screen is black) and the audio continues working.

Note: After 200 OK, asterisk does not forward the H.264 video RTP

I attach screenshots.

Can you give me a hand?

Very thanks,

NAT problem router.