Hikvision intercom only audio

Asterisk Asterisk Endpoints
1

Hello,
First of all sorry for my english, and my little knowledge about this topic.

I have a Hikvision IP intercom, and I plan to use it with Asterisk and Linhome app.
When I use a Linhome free account (XXX@sip.linhome.org) everything works fine, the intercom can call my app, and my app can call the intercom. All with sound and video.

However, I would like to make it local. So I though about using Asterisk. I just made two extensions, one for the intercom and other for the Linhome app. Now, when the intercom calls the app it works well, however if the app calls the intercom there is only audio no video.
I dont know where to begin to debug it to find the problem, I would appreciate any advice.

Thank you in advance.

2

I’ve been working on this for a few months. It looks like the Hikvision Doorphones h264 isnt matching up with asterisk. I’m trying to accomplish Hikvision Doorphones - Cisco 8865 videophones so far audio only

This is chatgpt’s analysis of the SIP exchange .

Analysis of SIP SDP Exchange

From your working Cisco-to-Cisco SIP SDP trace, we can see:

  • Audio Negotiation:
    • Audio is being negotiated correctly, using multiple codecs (PCMU, PCMA, G722, G729, iLBC, AMR-WB, etc.).
    • a=sendrecv confirms both send and receive directions.
  • Video Negotiation:
    • m=video 16472 RTP/AVP 112 111 110 (This means video is being offered on port 16472 with payloads 112, 111, 110).
    • H.264 Payloads:
      • a=rtpmap:112 H264/90000
      • a=rtpmap:111 H264/90000
      • a=rtpmap:110 H264/90000
    • Packetization Mode Handling:
      • a=fmtp:112 profile-level-id=640c16;packetization-mode=1
      • a=fmtp:111 profile-level-id=428016;packetization-mode=1
      • a=fmtp:110 profile-level-id=428016;packetization-mode=0

Comparing with Hikvision (Failed Call)

In your Hikvision-to-Cisco call, the SDP showed:

  • m=video 0 RTP/AVP 96
    • Port 0 means no video stream was established.
  • The Cisco phone is expecting:
    • Packetization mode 1 or 0.
    • H.264 with dynamic payloads (112, 111, 110).
    • Max MBPS, FS settings (max-fs=3600; max-mbps=108000).

Possible Causes and Fixes

1. Hikvision Not Sending Video Properly

  • Ensure the Hikvision device is actually offering video.
  • In the Hikvision Web UI, go to:
    • SIP → Advanced Settings → Video Settings
    • Enable Force H.264.
    • Set Packetization Mode to 1.
    • Manually define Dynamic Payload Type = 112 or 111 to match Cisco.

2. FreePBX/Asterisk Not Handling H.264 Correctly

  • Enable video support in FreePBX:
    • In Asterisk SIP Settings → General SIP Settings
3

Please provide the actual SDP, not just ChatGPT’s analysis of it.

4

However, when I call from the intercom to the app it sends video…

5

Hi, here is my log (well, the part I think matters…)

[2025-02-12 22:27:38] VERBOSE[60919] res_pjsip_logger.c: <--- Transmitting SIP request (603 bytes) to UDP:192.168.1.32:48485 --->
UPDATE sip:222222@192.168.1.32:48485;gr=urn:uuid:d1f121fc-05f3-49ba-be08-97cbf697c81e SIP/2.0
Via: SIP/2.0/UDP 192.168.1.24:5060;rport;branch=z9hG4bKPjda10a6ec-88ce-433e-ab85-9021d6d0407d
From: <sip:30003@192.168.1.24>;tag=785c1ab2-e1b9-41ce-92fe-79d1b7a0a28e
To: <sip:222222@sip.linphone.org>;tag=HxXo9Huwj
Contact: <sip:192.168.1.24:5060>
Call-ID: d8SiipIpwf
CSeq: 22802 UPDATE
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800
Min-SE: 90
P-Asserted-Identity: "Door" <sip:30003@192.168.1.24>
Max-Forwards: 70
User-Agent: FPBX-17.0.19.23(21.6.0)
Content-Length:  0


[2025-02-12 22:27:38] VERBOSE[128007] res_pjsip_logger.c: <--- Transmitting SIP request (1363 bytes) to UDP:192.168.1.165:5060 --->
INVITE sip:30003@192.168.1.165:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.24:5060;rport;branch=z9hG4bKPjeee6906e-1865-4d97-92f4-502e8df6c435
From: "APP" <sip:222222@192.168.1.24>;tag=86c65312-ef6c-434a-ac1b-73ae1a177fac
To: <sip:30003@192.168.1.165>
Contact: <sip:asterisk@192.168.1.24:5060>
Call-ID: 11a3b3bc-4d8e-4eea-af44-b464cdd48c7f
CSeq: 14046 INVITE
Allow: OPTIONS, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, MESSAGE, INFO, REFER
Supported: 100rel, timer, replaces, norefersub, histinfo
Session-Expires: 1800
Min-SE: 90
P-Asserted-Identity: "APP" <sip:222222@192.168.1.24>
Max-Forwards: 70
User-Agent: FPBX-17.0.19.23(21.6.0)
Content-Type: application/sdp
Content-Length:   611

v=0
o=- 1045396850 1045396850 IN IP4 192.168.1.24
s=Asterisk
c=IN IP4 192.168.1.24
t=0 0
m=audio 10688 RTP/AVP 107 0 8 3 111 9 101 102
a=rtpmap:107 opus/48000/2
a=fmtp:107 useinbandfec=1
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/48000
a=fmtp:101 0-16
a=rtpmap:102 telephone-event/8000
a=fmtp:102 0-16
a=ptime:20
a=maxptime:20
a=sendrecv
m=video 14668 RTP/AVP 99 104 109
a=rtpmap:99 H264/90000
a=fmtp:99 profile-level-id=42801F
a=rtpmap:104 MP4V-ES/90000
a=rtpmap:109 H265/90000
a=sendrecv

[2025-02-12 22:27:38] VERBOSE[1064] res_pjsip_logger.c: <--- Received SIP response (411 bytes) from UDP:192.168.1.32:48485 --->
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 192.168.1.24:5060;rport;branch=z9hG4bKPjda10a6ec-88ce-433e-ab85-9021d6d0407d
From: <sip:30003@192.168.1.24>;tag=785c1ab2-e1b9-41ce-92fe-79d1b7a0a28e
To: <sip:222222@sip.linphone.org>;tag=HxXo9Huwj
Call-ID: d8SiipIpwf
CSeq: 22802 UPDATE
User-Agent: Linhome/1.3 (Xiaomi) LinphoneSDK/5.3.105-pre.5+f4f904ed (pipelines/89778)
Supported: replaces, outbound, gruu, path


[2025-02-12 22:27:38] VERBOSE[1064] res_pjsip_logger.c: <--- Received SIP response (369 bytes) from UDP:192.168.1.165:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.24:5060;rport=5060;branch=z9hG4bKPjeee6906e-1865-4d97-92f4-502e8df6c435;received=192.168.1.24
From: "APP" <sip:222222@192.168.1.24>;tag=86c65312-ef6c-434a-ac1b-73ae1a177fac
To: <sip:30003@192.168.1.165>
Call-ID: 11a3b3bc-4d8e-4eea-af44-b464cdd48c7f
CSeq: 14046 INVITE
Server: YATE/5.5.0
Content-Length: 0


[2025-02-12 22:27:38] VERBOSE[1064] res_pjsip_logger.c: <--- Received SIP response (776 bytes) from UDP:192.168.1.165:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.24:5060;rport=5060;branch=z9hG4bKPjeee6906e-1865-4d97-92f4-502e8df6c435;received=192.168.1.24
From: "APP" <sip:222222@192.168.1.24>;tag=86c65312-ef6c-434a-ac1b-73ae1a177fac
To: <sip:30003@192.168.1.165>;tag=1362895963
Call-ID: 11a3b3bc-4d8e-4eea-af44-b464cdd48c7f
CSeq: 14046 INVITE
Server: YATE/5.5.0
Contact: <sip:30003@192.168.1.165:5060>
Allow: ACK, INVITE, BYE, CANCEL, MESSAGE, OPTIONS, INFO
Content-Type: application/sdp
Content-Length: 265

v=0
o=FG1807002 0 0 IN IP4 192.168.1.165
s=Talk session
c=IN IP4 192.168.1.165
t=0 0
m=audio 9654 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=sendrecv
m=video 9856 RTP/AVP 96
a=rtpmap:96 H264/90000
a=fmtp:96 profile-level-id=42801F;packetization-mode=1
a=sendonly

[2025-02-12 22:27:38] VERBOSE[80824] res_pjsip_logger.c: <--- Transmitting SIP request (412 bytes) to UDP:192.168.1.165:5060 --->
ACK sip:30003@192.168.1.165:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.24:5060;rport;branch=z9hG4bKPj79ba8459-d13b-4806-ba88-10c1b9a0608e
From: "APP" <sip:222222@192.168.1.24>;tag=86c65312-ef6c-434a-ac1b-73ae1a177fac
To: <sip:30003@192.168.1.165>;tag=1362895963
Call-ID: 11a3b3bc-4d8e-4eea-af44-b464cdd48c7f
CSeq: 14046 ACK
Max-Forwards: 70
User-Agent: FPBX-17.0.19.23(21.6.0)
Content-Length:  0```

Please, tell me if you need some other information let me know. Thank you very much.
6

You set up an outgoing call, with H.264 video, incoming only.

There is no corresponding incoming call, or second outgoing call, and no dialplan execution has been logged, the latter probably because verbosity is too low to see it.

There is part of a call involving “door”, but no SDP is exchanged.

7

Hello again, sorry I think now there is more information in the log. Thank you again for your time.

Here is the log. I am a new user so I can’t attach files.

8

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