Frustrated: Cant get Early Media to work!

FreePBX Endpoints
1

Struggling with this for a long time already, running freepbx with asterisk 16.17

i have a doorstation and some linphone clients, when i press the call button, i only see video AFTER i pickup, i want to see it before i pickup
I have tested other PBX too like fusion/freeswitch, then it works… so i know its not a client side issue, but a PBX issue
its a simple setup, no trunks, all in on local lan network…, created some chan_sip extensions, enabled video, enabled correct codecs…
tried also enabling these settings below, still a no go…

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image922×141 5.19 KB

what am i missing?

thnx, much appreciated!!!

here is full log file : Dropbox - freepbx log linphone door+video - Simplify your life

INVITE sip:[email protected]:40643;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.0.174:5160;branch=z9hG4bK50cf6913;rport
Max-Forwards: 70
From: "2000" <sip:[email protected]:5160>;tag=as36fbff33
To: <sip:[email protected]:40643;transport=udp>
Contact: <sip:[email protected]:5160>
Call-ID: [email protected]:5160
CSeq: 102 INVITE
User-Agent: FPBX-15.0.17.34(16.17.0)
Date: Thu, 14 Oct 2021 17:21:47 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
P-Asserted-Identity: "2000" <sip:[email protected]>
Content-Type: application/sdp
Content-Length: 470

v=0
o=root 957584624 957584624 IN IP4 192.168.0.174
s=Asterisk PBX 16.17.0
c=IN IP4 192.168.0.174
b=CT:384
t=0 0
m=audio 12778 RTP/AVP 0 8 3 111 9 101
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/8000
a=fmtp:101 0-16
a=maxptime:150
a=sendrecv
m=video 19822 RTP/AVP 99
a=rtpmap:99 H264/90000
a=fmtp:99 packetization-mode=1;profile-level-id=4D001F
a=sendrecv

---
[2021-10-14 17:21:47] VERBOSE[31904][C-00000003] app_dial.c: Called SIP/2002
[2021-10-14 17:21:47] VERBOSE[31904][C-00000003] chan_sip.c: 
<--- Transmitting (NAT) to 192.168.0.70:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.0.70:5060;branch=z9hG4bK574991794;received=192.168.0.70;rport=5060
From: "2000" <sip:[email protected]>;tag=78128501
To: <sip:[email protected]:5160>;tag=as44581885
Call-ID: [email protected]
CSeq: 165 INVITE
Server: FPBX-15.0.17.34(16.17.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:[email protected]:5160>
P-Asserted-Identity: "2002" <sip:[email protected]>
Content-Length: 0


<------------>
[2021-10-14 17:21:47] VERBOSE[2267] chan_sip.c: Retransmitting #1 (NAT) to 192.168.0.168:40643:
INVITE sip:[email protected]:40643;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.0.174:5160;branch=z9hG4bK50cf6913;rport
Max-Forwards: 70
From: "2000" <sip:[email protected]:5160>;tag=as36fbff33
To: <sip:[email protected]:40643;transport=udp>
Contact: <sip:[email protected]:5160>
Call-ID: [email protected]:5160
CSeq: 102 INVITE
User-Agent: FPBX-15.0.17.34(16.17.0)
Date: Thu, 14 Oct 2021 17:21:47 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
P-Asserted-Identity: "2000" <sip:[email protected]>
Content-Type: application/sdp
Content-Length: 470

v=0
o=root 957584624 957584624 IN IP4 192.168.0.174
s=Asterisk PBX 16.17.0
c=IN IP4 192.168.0.174
b=CT:384
t=0 0
m=audio 12778 RTP/AVP 0 8 3 111 9 101
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/8000
a=fmtp:101 0-16
a=maxptime:150
a=sendrecv
m=video 19822 RTP/AVP 99
a=rtpmap:99 H264/90000
a=fmtp:99 packetization-mode=1;profile-level-id=4D001F
a=sendrecv

---
[2021-10-14 17:21:47] VERBOSE[2267] chan_sip.c: 
<--- SIP read from UDP:192.168.0.168:40643 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.174:5160;branch=z9hG4bK50cf6913;rport
From: "2000" <sip:[email protected]:5160>;tag=as36fbff33
To: <sip:[email protected]:40643;transport=udp>
Call-ID: [email protected]:5160
CSeq: 102 INVITE

<------------->
[2021-10-14 17:21:47] VERBOSE[2267] chan_sip.c: --- (6 headers 0 lines) ---
[2021-10-14 17:21:47] VERBOSE[2267] chan_sip.c: 
<--- SIP read from UDP:192.168.0.168:40643 --->
SIP/2.0 183 Session progress
Via: SIP/2.0/UDP 192.168.0.174:5160;branch=z9hG4bK50cf6913;rport
From: "2000" <sip:[email protected]:5160>;tag=as36fbff33
To: <sip:[email protected]:40643;transport=udp>;tag=dyuK~xp
Call-ID: [email protected]:5160
CSeq: 102 INVITE
User-Agent: Linphone/4.5.3 (OnePlus 6T) LinphoneSDK/5.0.31 (tags/5.0.31^0)
Supported: replaces, outbound, gruu
Content-Type: application/sdp
Content-Length: 254

v=0
o=2002 1937 3737 IN IP4 192.168.0.168
s=Talk
c=IN IP4 192.168.0.168
t=0 0
m=audio 7078 RTP/AVP 0 8 101
a=rtpmap:101 telephone-event/8000
m=video 9078 RTP/AVP 99
a=rtpmap:99 H264/90000
a=fmtp:99 profile-level-id=42801F; packetization-mode=1
<------------->
[2021-10-14 17:21:47] VERBOSE[2267] chan_sip.c: --- (10 headers 10 lines) ---
[2021-10-14 17:21:47] VERBOSE[2267][C-00000003] sip/route.c: sip_route_dump: no route/path
[2021-10-14 17:21:47] VERBOSE[2267][C-00000003] chan_sip.c: Got SDP version 3737 and unique parts [2002 1937 IN IP4 192.168.0.168]
[2021-10-14 17:21:47] VERBOSE[2267][C-00000003] chan_sip.c: Found RTP audio format 0
[2021-10-14 17:21:47] VERBOSE[2267][C-00000003] chan_sip.c: Found RTP audio format 8
[2021-10-14 17:21:47] VERBOSE[2267][C-00000003] chan_sip.c: Found RTP audio format 101
[2021-10-14 17:21:47] VERBOSE[2267][C-00000003] chan_sip.c: Found audio description format telephone-event for ID 101
[2021-10-14 17:21:47] VERBOSE[2267][C-00000003] chan_sip.c: Found RTP video format 99
[2021-10-14 17:21:47] VERBOSE[2267][C-00000003] chan_sip.c: Found video description format H264 for ID 99
[2021-10-14 17:21:47] VERBOSE[2267][C-00000003] chan_sip.c: Capabilities: us - (ulaw|alaw|gsm|g726|g722|h264|mpeg4), peer - audio=(ulaw|alaw)/video=(h264)/text=(nothing), combined - (ulaw|alaw|h264)
[2021-10-14 17:21:47] VERBOSE[2267][C-00000003] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
[2021-10-14 17:21:47] VERBOSE[2267][C-00000003] chan_sip.c: Peer audio RTP is at port 192.168.0.168:7078
[2021-10-14 17:21:47] VERBOSE[2267][C-00000003] chan_sip.c: Peer video RTP is at port 192.168.0.168:9078
[2021-10-14 17:21:47] VERBOSE[31904][C-00000003] app_dial.c: SIP/2002-00000005 is making progress passing it to SIP/2000-00000004
[2021-10-14 17:21:48] VERBOSE[2267] chan_sip.c: 
<--- SIP read from UDP:192.168.0.168:40643 --->
SIP/2.0 183 Session progress
Via: SIP/2.0/UDP 192.168.0.174:5160;branch=z9hG4bK50cf6913;rport
From: "2000" <sip:[email protected]:5160>;tag=as36fbff33
To: <sip:[email protected]:40643;transport=udp>;tag=dyuK~xp
Call-ID: [email protected]:5160
CSeq: 102 INVITE
User-Agent: Linphone/4.5.3 (OnePlus 6T) LinphoneSDK/5.0.31 (tags/5.0.31^0)
Supported: replaces, outbound, gruu
Content-Type: application/sdp
Content-Length: 254

v=0
o=2002 1937 3737 IN IP4 192.168.0.168
s=Talk
c=IN IP4 192.168.0.168
t=0 0
m=audio 7078 RTP/AVP 0 8 101
a=rtpmap:101 telephone-event/8000
m=video 9078 RTP/AVP 99
a=rtpmap:99 H264/90000
a=fmtp:99 profile-level-id=42801F; packetization-mode=1
<------------->
[2021-10-14 17:21:48] VERBOSE[2267] chan_sip.c: --- (10 headers 10 lines) ---
[2021-10-14 17:21:48] VERBOSE[2267][C-00000003] sip/route.c: sip_route_dump: no route/path
[2021-10-14 17:21:48] VERBOSE[2267][C-00000003] chan_sip.c: Comparing SDP version 3737 -> 3737 and unique parts [2002 1937 IN IP4 192.168.0.168] -> [2002 1937 IN IP4 192.168.0.168]
[2021-10-14 17:21:48] VERBOSE[31904][C-00000003] app_dial.c: SIP/2002-00000005 is making progress passing it to SIP/2000-00000004
[2021-10-14 17:21:49] VERBOSE[2267] chan_sip.c: 
<--- SIP read from UDP:192.168.0.168:40643 --->
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 192.168.0.174:5160;branch=z9hG4bK50cf6913;rport
From: "2000" <sip:[email protected]:5160>;tag=as36fbff33
To: <sip:[email protected]:40643;transport=udp>;tag=dyuK~xp
Call-ID: [email protected]:5160
CSeq: 102 INVITE
User-Agent: Linphone/4.5.3 (OnePlus 6T) LinphoneSDK/5.0.31 (tags/5.0.31^0)
Supported: replaces, outbound, gruu
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO, PRACK, UPDATE
Contact: <sip:[email protected]:40643;transport=udp>;expires=3600;+sip.instance="<urn:uuid:e5f09345-a52b-0097-aa78-e2b34b9a3ea0>";+org.linphone.specs="lime"
Content-Type: application/sdp
Content-Length: 254

v=0
o=2002 1937 3738 IN IP4 192.168.0.168
s=Talk
c=IN IP4 192.168.0.168
t=0 0
m=audio 7078 RTP/AVP 0 8 101
a=rtpmap:101 telephone-event/8000
m=video 9078 RTP/AVP 99
a=rtpmap:99 H264/90000
a=fmtp:99 profile-level-id=42801F; packetization-mode=1
<------------->
[2021-10-14 17:21:49] VERBOSE[2267] chan_sip.c: --- (12 headers 10 lines) ---
[2021-10-14 17:21:49] VERBOSE[2267][C-00000003] chan_sip.c: Comparing SDP version 3737 -> 3738 and unique parts [2002 1937 IN IP4 192.168.0.168] -> [2002 1937 IN IP4 192.168.0.168]
[2021-10-14 17:21:49] VERBOSE[2267][C-00000003] chan_sip.c: Found RTP audio format 0
[2021-10-14 17:21:49] VERBOSE[2267][C-00000003] chan_sip.c: Found RTP audio format 8
[2021-10-14 17:21:49] VERBOSE[2267][C-00000003] chan_sip.c: Found RTP audio format 101
[2021-10-14 17:21:49] VERBOSE[2267][C-00000003] chan_sip.c: Found audio description format telephone-event for ID 101
[2021-10-14 17:21:49] VERBOSE[2267][C-00000003] chan_sip.c: Found RTP video format 99
[2021-10-14 17:21:49] VERBOSE[2267][C-00000003] chan_sip.c: Found video description format H264 for ID 99
[2021-10-14 17:21:49] VERBOSE[2267][C-00000003] chan_sip.c: Capabilities: us - (ulaw|alaw|gsm|g726|g722|h264|mpeg4), peer - audio=(ulaw|alaw)/video=(h264)/text=(nothing), combined - (ulaw|alaw|h264)
[2021-10-14 17:21:49] VERBOSE[2267][C-00000003] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
[2021-10-14 17:21:49] VERBOSE[2267][C-00000003] chan_sip.c: Peer audio RTP is at port 192.168.0.168:7078
[2021-10-14 17:21:49] VERBOSE[2267][C-00000003] chan_sip.c: Peer video RTP is at port 192.168.0.168:9078
[2021-10-14 17:21:49] VERBOSE[2267][C-00000003] sip/route.c: sip_route_dump: route/path hop: <sip:[email protected]:40643;transport=udp>
[2021-10-14 17:21:49] VERBOSE[2267][C-00000003] chan_sip.c: Transmitting (NAT) to 192.168.0.168:40643:
ACK sip:[email protected]:40643;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.0.174:5160;branch=z9hG4bK44f42b66;rport
Max-Forwards: 70
From: "2000" <sip:[email protected]:5160>;tag=as36fbff33
To: <sip:[email protected]:40643;transport=udp>;tag=dyuK~xp
Contact: <sip:[email protected]:5160>
Call-ID: [email protected]:5160
CSeq: 102 ACK
User-Agent: FPBX-15.0.17.34(16.17.0)
Content-Length: 0


---
[2021-10-14 17:21:49] VERBOSE[31904][C-00000003] app_dial.c: SIP/2002-00000005 answered SIP/2000-00000004
[2021-10-14 17:21:49] VERBOSE[31904][C-00000003] chan_sip.c: Audio is at 15440
[2021-10-14 17:21:49] VERBOSE[31904][C-00000003] chan_sip.c: Video is at 192.168.0.174:16078
[2021-10-14 17:21:49] VERBOSE[31904][C-00000003] chan_sip.c: Adding codec ulaw to SDP
[2021-10-14 17:21:49] VERBOSE[31904][C-00000003] chan_sip.c: Adding video codec h264 to SDP
[2021-10-14 17:21:49] VERBOSE[31904][C-00000003] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
[2021-10-14 17:21:49] VERBOSE[31904][C-00000003] chan_sip.c: 
<--- Reliably Transmitting (NAT) to 192.168.0.70:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.70:5060;branch=z9hG4bK574991794;received=192.168.0.70;rport=5060
From: "2000" <sip:[email protected]>;tag=78128501
To: <sip:[email protected]:5160>;tag=as44581885
Call-ID: [email protected]
CSeq: 165 INVITE
Server: FPBX-15.0.17.34(16.17.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:[email protected]:5160>
P-Asserted-Identity: "2002" <sip:[email protected]>
Content-Type: application/sdp
Content-Length: 370

v=0
o=root 1477218312 1477218312 IN IP4 192.168.0.174
s=Asterisk PBX 16.17.0
c=IN IP4 192.168.0.174
b=CT:384
t=0 0
m=audio 15440 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:150
a=sendrecv
m=video 16078 RTP/AVP 96
a=rtpmap:96 H264/90000
a=fmtp:96 packetization-mode=1;profile-level-id=4D001F
a=sendrecv
2

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