SIP Video Call Support

Asterisk can relay video streams between endpoints. Video support requires enabling the right codecs on each endpoint and ensuring both sides negotiate a common video codec. Asterisk acts as a media relay (pass-through) for video, it does not transcode video.

PJSIP Endpoint Template (pjsip.conf)

; Base template for video-capable endpoints
[endpoint-video](!)
type = endpoint
transport = transport-udp
allow = !all,ulaw,alaw,opus,h264,vp8
max_video_streams = 1

Video Endpoint Examples (pjsip.conf)

; Hardware SIP phone with H.264 support
[1001](endpoint-internal,endpoint-video)
auth = 1001-auth
aors = 1001
callerid = Alice <1001>

; Softphone with multiple video codecs
[1002](endpoint-internal,endpoint-video)
auth = 1002-auth
aors = 1002
callerid = Bob <1002>
allow = !all,ulaw,opus,vp9,vp8,h264

WebRTC Video Endpoint (pjsip.conf)

[2001](endpoint-internal)
transport = transport-wss
auth = 2001-auth
aors = 2001
allow = !all,opus,vp9,vp8,h264
dtls_auto_generate_cert = yes
webrtc = yes
max_video_streams = 1

Video Codec Comparison

Dialplan for Video Calls

[internal]
; Video calls work with standard Dial() -- no special dialplan needed.
; Both endpoints must share at least one video codec.
exten => _1XXX,1,Dial(PJSIP/${EXTEN},30)
 same => n,Hangup()

Verify Video Codec Support

*CLI> core show codecs video
  Codec   Frames/Pkt  Bytes/Frame  Codec
  h261    (20)         -            H.261 video
  h263    (20)         -            H.263 video
  h263p   (20)         -            H.263+ video
  h264    (20)         -            H.264 video
  vp8     (20)         -            VP8 video
  vp9     (20)         -            VP9 video

*CLI> pjsip show endpoint 1001
  ...
  allow: (ulaw|alaw|opus|h264|vp8)
  ...

How it works

1. Pass-through only: Asterisk does not transcode video. It relays video packets between endpoints as-is. Both sides must negotiate a common video codec during the SDP offer/answer exchange, or the video stream will not be established (audio still works).
2. allow codec list: Video codecs are added to the same allow line as audio codecs. Asterisk uses the order to set codec preference during negotiation. List preferred codecs first.
3. max_video_streams: Controls how many simultaneous video streams an endpoint can have. Set to 1 for standard point-to-point video calls. Set higher for multiparty video conferencing via ConfBridge.
4. WebRTC video: The webrtc = yes shorthand enables DTLS-SRTP, ICE, AVPF, and RTCP multiplexing. VP8 and VP9 are the standard video codecs for browser-based WebRTC clients. H.264 is also supported by most browsers.
5. Codec negotiation: If endpoint A offers h264,vp8 and endpoint B supports only vp8, the call will use VP8 for video. If there is no overlap, the video portion of the call is dropped but audio proceeds normally.

Tips

• Video calls consume significantly more bandwidth than audio-only calls. Expect 500 kbps to 2+ Mbps per video stream depending on resolution and codec.
• Test video between two softphones first. Linphone and MicroSIP both support H.264 and are free.
• For ConfBridge video conferences, set video_mode = follow_talker in the bridge profile to automatically switch the video feed to the active speaker.
• Some SIP phones only work with one video codec enabled. If video fails, try restricting to a single video codec on that endpoint.
• The codec_h264 module provides H.264 pass-through. It does not require a license since Asterisk is not encoding or decoding the video.
• Use rtp set debug on to verify that video RTP packets are flowing between endpoints during a call.