Direct media=yes not working

Asterisk Asterisk SIP
1

Directmedia=yes but the media is still passing through asterisk/

2

My dialplan:

exten => _X.,1,NoOp(New Outgoing call from ${CALLERID(num)} to ${EXTEN:0})
same => n, Set(CALLERLEN=${LEN(${CALLERID(num)})})
same => n, GotoIf($[${CALLERLEN}!=6]?rotate)
same => n, Set(CRMSTATE=${DEVICE_STATE(Custom:Crm${CALLERID(num)})})
same => n, GotoIf($[$[${CRMSTATE}!=NOT_INUSE] & $[${CRMSTATE}!=BUSY]]?Kill,crmoffline,1)
same => n(call),Dial(SIP/${EXTEN:0},${RINGTIME})

sip.conf
id : 1072
name : 211002
callerid : xxxxxx
secret : xxxx
mailbox : NULL
accountcode : NULL
context : Internal
amaflags : NULL
callgroup : NULL
defaultip : NULL
dtmfmode : none
fromuser : NULL
fromdomain : NULL
fullcontact : sip:u0as5k9e@192.0.2.54;transport=wss
host : dynamic
insecure : NULL
language : NULL
md5secret : NULL
nat : no
mask : NULL
pickupgroup : NULL
restrictcid : NULL
rtptimeout : 0
type : friend
disallow : all
allow : opus,vp8
musiconhold : NULL
regseconds : 1758543326
useragent : xxxxx
regserver : (empty)
transport : wss
lastms : 9
port : 14857
avpf : yes
encryption : yes
icesupport : yes
rtcp_mux : yes
dtlsenable : yes
dtlsverify : yes
dtlscertfile : xxxxxx
dtlscafile : xxxxxx
dtlssetup : actpass
videosupport : yes
force_avpf : yes
realm : (empty)
srvlookup : no
mohsuggest : default
parkinglot : default
allowguest : no
alwaysauthreject : yes
maxcallbitrate : 5120
ignoreregexpire : no
notifyhold : yes
notifyringing : yes
callcounter : yes
progressinband : yes
tos_sip : af42
tos_audio : ef
cos_sip : 3
cos_audio : 5
rtpkeepalive : 60
tcpenable : no
tlsenable : no
websocket_enabled : yes
directmedia : yes
subscribecontext : subscriptions
defaultuser : 211002
allowsubscribe : yes
outofcall_message_context : textmessages
auth_message_requests : no
accept_outofcall_messages : yes
udpbindaddr : 0.0.0.0:5060
externip : (empty)
qualify : yes
call-limit : 1
allowheaders : X-sourceType
dtlsprivatekey : xxxxx
rewrite_contact : yes
localnet : (empty)
bridge_native_rtp : yes
force_rport : no
external_media_address : xxxx
media_address : xxxx
comedia : no
native_bridge_rtp : yes
use_native_rtp_bridge : yes
use_native_bridge_rtp : yes
ipaddr : xxxxx

3

If using WebRTC, or media encryption in general, then direct media is not supported.

4

Can anything be done ro improve video quality?

5

Under chan_sip? No. It doesn’t support some of the additional SDP functionality that chan_pjsip does, some of which are for bitrate and bandwidth.

6

As well, the clients are what consume that information/gauge it in order to determine bitrate and bandwidth to adjust the video quality. Asterisk negotiates it and passes it around, as well as adding some details into the RTP packets so it works.

7

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