Video works locally but not externally with NAT

Asterisk Asterisk Support
1

Hi there,

I have Asterisk 13 installed on my Raspberry Pi. On my laptop at home, I use Jitsi as the SIP soft phone. When I place a call to my front door intercom (Dahua VTO2000A using extension 8001), the call includes video and I am able to see what’s happening at the front door.

When I use my laptop at work, also with Jitsi installed (using extension 8006), the same call does not include video.

From what I can tell, the only real difference between the two scenarios is local network vs. external NAT.

Below are the settings from sip.conf, and further below is the SIP debug.

Any ideas?

Thanks!
Pete

[general]
bindport=5060
bindaddr=0.0.0.0
context=default
allowguest=no
alwaysauthreject=yes
srvlookup=yes
udpbindaddr=0.0.0.0
tcpenable=no
localnet=192.168.1.0/255.255.255.0
externhost=my.own.domain
externrefresh=120
nat=force_rport,comedia

[soft-phone](!)
type=friend
context=internal
host=dynamic
dtmfmode=rfc2833
qualify=yes
directmedia=no
canreinvite=no
dtmfmode=rfc2833
secret=*****************
videosupport=yes
disallow=all
allow=ulaw
allow=alaw
allow=h263p
allow=h263
allow=vp8
allow=h264

[8001](soft-phone)
disallow=all
allow=ulaw
allow=h264
nat=force_rport,comedia

[8006](soft-phone)
nat=force_rport,comedia

Here is the SIP debug:

<--- SIP read from UDP:211.27.124.218:12889 --->
INVITE sip:8001@my.own.domain SIP/2.0
Call-ID: 671d7eb3169a29b318535e9710760c3d@0:0:0:0:0:0:0:0
CSeq: 2 INVITE
From: "8006" <sip:8006@my.own.domain>;tag=1b3e70a
To: <sip:8001@my.own.domain>
Max-Forwards: 70
Contact: "8006" <sip:8006@211.27.124.218:12889;transport=udp;registering_acc=my_own_domain>
User-Agent: Jitsi2.8.5426Windows 7
Content-Type: application/sdp
Via: SIP/2.0/UDP 211.27.124.218:12889;branch=z9hG4bK-333636-a6025ae90a84f730bb88b846d9e6c7c5
Authorization: Digest username="8006",realm="asterisk",nonce="3cb4a102",uri="sip:8001@my.own.domain",response="6d463a91bda7c584499c21ff6a45a2b2",algorithm=MD5
Content-Length: 1054

v=0
o=8006-jitsi.org 0 0 IN IP4 211.27.124.218
s=-
c=IN IP4 211.27.124.218
t=0 0
m=audio 9312 RTP/AVP 96 97 98 9 100 102 0 8 103 3 104 4 101
a=rtpmap:96 opus/48000/2
a=fmtp:96 usedtx=1
a=rtpmap:97 SILK/24000
a=rtpmap:98 SILK/16000
a=rtpmap:9 G722/8000
a=rtpmap:100 speex/32000
a=rtpmap:102 speex/16000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:103 iLBC/8000
a=rtpmap:3 GSM/8000
a=rtpmap:104 speex/8000
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no;bitrate=6.3
a=rtpmap:101 telephone-event/8000
a=extmap:1 urn:ietf:params:rtp-hdrext:csrc-audio-level
a=extmap:2 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=rtcp-xr:voip-metrics
m=video 30688 RTP/AVP 105 99 106 107
a=recvonly
a=rtpmap:105 H264/90000
a=fmtp:105 profile-level-id=4DE01f;packetization-mode=1
a=imageattr:105 send * recv [x=[0-1920],y=[0-1080]]
a=rtpmap:99 H264/90000
a=fmtp:99 profile-level-id=4DE01f
a=imageattr:99 send * recv [x=[0-1920],y=[0-1080]]
a=rtpmap:106 H263-1998/90000
a=fmtp:106 CUSTOM=1920,1080,2;VGA=2;CIF=1;QCIF=1
a=rtpmap:107 VP8/90000
<------------->
--- (12 headers 35 lines) ---
Sending to 211.27.124.218:12889 (NAT)
Using INVITE request as basis request - 671d7eb3169a29b318535e9710760c3d@0:0:0:0:0:0:0:0
Found peer '8006' for '8006' from 211.27.124.218:12889
  == Using SIP VIDEO CoS mark 6
  == Using SIP RTP CoS mark 5
Found RTP audio format 96
Found RTP audio format 97
Found RTP audio format 98
Found RTP audio format 9
Found RTP audio format 100
Found RTP audio format 102
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 103
Found RTP audio format 3
Found RTP audio format 104
Found RTP audio format 4
Found RTP audio format 101
Found audio description format opus for ID 96
Found unknown media description format SILK for ID 97
Found unknown media description format SILK for ID 98
Found audio description format G722 for ID 9
Found audio description format speex for ID 100
Found audio description format speex for ID 102
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format iLBC for ID 103
Found audio description format GSM for ID 3
Found audio description format speex for ID 104
Found audio description format G723 for ID 4
Found audio description format telephone-event for ID 101
Found RTP video format 105
Found RTP video format 99
Found RTP video format 106
Found RTP video format 107
Found video description format H264 for ID 105
Found video description format H264 for ID 99
Found video description format H263-1998 for ID 106
Found video description format VP8 for ID 107
Capabilities: us - (ulaw|h264), peer - audio=(ulaw|gsm|g723|alaw|g722|opus|speex32|speex16|ilbc|speex)/video=(h264|h263p|vp8)/text=(nothing), combined - (ulaw|h264)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 211.27.124.218:9312
Peer video RTP is at port 211.27.124.218:30688
Looking for 8001 in internal (domain my.own.domain)
sip_route_dump: route/path hop: <sip:8006@211.27.124.218:12889;transport=udp;registering_acc=my_own_domain>

<--- Transmitting (NAT) to 211.27.124.218:12889 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 211.27.124.218:12889;branch=z9hG4bK-333636-a6025ae90a84f730bb88b846d9e6c7c5;received=211.27.124.218;rport=12889
From: "8006" <sip:8006@my.own.domain>;tag=1b3e70a
To: <sip:8001@my.own.domain>
Call-ID: 671d7eb3169a29b318535e9710760c3d@0:0:0:0:0:0:0:0
CSeq: 2 INVITE
Server: Asterisk PBX 13.12.2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:8001@137.147.146.218:5060>
Content-Length: 0


<------------>
    -- Executing [8001@internal:1] Dial("SIP/8006-00000002", "SIP/8001") in new stack
  == Using SIP VIDEO CoS mark 6
  == Using SIP RTP CoS mark 5
Audio is at 13582
Video is at 192.168.1.250:15102
Adding codec ulaw to SDP
Adding video codec h264 to SDP
Reliably Transmitting (NAT) to 192.168.1.110:5060:
INVITE sip:8001@192.168.1.110:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.250:5060;branch=z9hG4bK082c9394;rport
Max-Forwards: 70
From: "8006" <sip:8006@192.168.1.250>;tag=as5d9d25c4
To: <sip:8001@192.168.1.110:5060>
Contact: <sip:8006@192.168.1.250:5060>
Call-ID: 20660e4f2971b5a3318c8aa24206e320@192.168.1.250:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 13.12.2
Date: Thu, 24 Nov 2016 00:16:46 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 293

v=0
o=root 2013624946 2013624946 IN IP4 192.168.1.250
s=Asterisk PBX 13.12.2
c=IN IP4 192.168.1.250
b=CT:384
t=0 0
m=audio 13582 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=maxptime:150
a=sendrecv
m=video 15102 RTP/AVP 99
a=rtpmap:99 H264/90000
a=fmtp:99 profile-level-id=4DE01F
a=sendrecv

---
    -- Called SIP/8001

<--- SIP read from UDP:192.168.1.110:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.250:5060;branch=z9hG4bK082c9394;rport=5060
From: "8006" <sip:8006@192.168.1.250>;tag=as5d9d25c4
To: <sip:8001@192.168.1.110:5060>
Call-ID: 20660e4f2971b5a3318c8aa24206e320@192.168.1.250:5060
CSeq: 102 INVITE
User-Agent: Dahua UAC/3.0 VTO2000A V1.200.1000.0
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---

<--- SIP read from UDP:192.168.1.110:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.1.250:5060;branch=z9hG4bK082c9394;rport=5060
From: "8006" <sip:8006@192.168.1.250>;tag=as5d9d25c4
To: <sip:8001@192.168.1.110:5060>;tag=1227522572
Call-ID: 20660e4f2971b5a3318c8aa24206e320@192.168.1.250:5060
CSeq: 102 INVITE
Contact: <sip:8001@192.168.1.110:5060>
User-Agent: Dahua UAC/3.0 VTO2000A V1.200.1000.0
MaxRingingTime: 30
MaxConnectingTime: 120
DependentInfo:
Content-Length: 0

<------------->
--- (12 headers 0 lines) ---
sip_route_dump: route/path hop: <sip:8001@192.168.1.110:5060>
    -- SIP/8001-00000003 is ringing

<--- Transmitting (NAT) to 211.27.124.218:12889 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 211.27.124.218:12889;branch=z9hG4bK-333636-a6025ae90a84f730bb88b846d9e6c7c5;received=211.27.124.218;rport=12889
From: "8006" <sip:8006@my.own.domain>;tag=1b3e70a
To: <sip:8001@my.own.domain>;tag=as05ce9243
Call-ID: 671d7eb3169a29b318535e9710760c3d@0:0:0:0:0:0:0:0
CSeq: 2 INVITE
Server: Asterisk PBX 13.12.2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:8001@137.147.146.218:5060>
Content-Length: 0


<------------>

<--- SIP read from UDP:192.168.1.110:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.250:5060;branch=z9hG4bK082c9394;rport=5060
From: "8006" <sip:8006@192.168.1.250>;tag=as5d9d25c4
To: <sip:8001@192.168.1.110:5060>;tag=1227522572
Call-ID: 20660e4f2971b5a3318c8aa24206e320@192.168.1.250:5060
CSeq: 102 INVITE
Contact: <sip:8001@192.168.1.110:5060>
User-Agent: Dahua UAC/3.0 VTO2000A V1.200.1000.0
Content-Type: application/sdp
Content-Length: 252

v=0
o=0 0 0 IN IP4 192.168.1.110
s=Dahua VT 1.5
c=IN IP4 192.168.1.110
t=0 0
m=video 20001 RTP/AVP 96
a=framerate:25.000000
a=rtpmap:96 H264/90000
a=sendrecv
m=audio 20000 RTP/AVP 97 0
a=rtpmap:97 PCM/16000
a=rtpmap:0 PCMU/8000
a=sendrecv
<------------->
--- (10 headers 13 lines) ---
Found RTP video format 96
Found video description format H264 for ID 96
Found RTP audio format 97
Found RTP audio format 0
Found unknown media description format PCM for ID 97
Found audio description format PCMU for ID 0
Capabilities: us - (ulaw|h264), peer - audio=(ulaw)/video=(h264)/text=(nothing), combined - (ulaw|h264)
Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x0 (nothing), combined - 0x0 (nothing)
Peer audio RTP is at port 192.168.1.110:20000
Peer video RTP is at port 192.168.1.110:20001
sip_route_dump: route/path hop: <sip:8001@192.168.1.110:5060>
Transmitting (NAT) to 192.168.1.110:5060:
ACK sip:8001@192.168.1.110:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.250:5060;branch=z9hG4bK4b561342;rport
Max-Forwards: 70
From: "8006" <sip:8006@192.168.1.250>;tag=as5d9d25c4
To: <sip:8001@192.168.1.110:5060>;tag=1227522572
Contact: <sip:8006@192.168.1.250:5060>
Call-ID: 20660e4f2971b5a3318c8aa24206e320@192.168.1.250:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 13.12.2
Content-Length: 0


---
    -- SIP/8001-00000003 answered SIP/8006-00000002
Audio is at 13874
Video is at 137.147.146.218:17168
Adding codec ulaw to SDP
Adding video codec h264 to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (NAT) to 211.27.124.218:12889 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 211.27.124.218:12889;branch=z9hG4bK-333636-a6025ae90a84f730bb88b846d9e6c7c5;received=211.27.124.218;rport=12889
From: "8006" <sip:8006@my.own.domain>;tag=1b3e70a
To: <sip:8001@my.own.domain>;tag=as05ce9243
Call-ID: 671d7eb3169a29b318535e9710760c3d@0:0:0:0:0:0:0:0
CSeq: 2 INVITE
Server: Asterisk PBX 13.12.2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:8001@137.147.146.218:5060>
Content-Type: application/sdp
Content-Length: 353

v=0
o=root 1399188747 1399188747 IN IP4 137.147.146.218
s=Asterisk PBX 13.12.2
c=IN IP4 137.147.146.218
b=CT:384
t=0 0
m=audio 13874 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 17168 RTP/AVP 99
a=rtpmap:99 H264/90000
a=fmtp:99 profile-level-id=4DE01F
a=sendrecv

<------------>
    -- Channel SIP/8001-00000003 joined 'simple_bridge' basic-bridge <aa3440d2-df4d-4a8e-84ca-acb8977c88b2>
    -- Channel SIP/8006-00000002 joined 'simple_bridge' basic-bridge <aa3440d2-df4d-4a8e-84ca-acb8977c88b2>
Retransmitting #1 (NAT) to 211.27.124.218:12889:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 211.27.124.218:12889;branch=z9hG4bK-333636-a6025ae90a84f730bb88b846d9e6c7c5;received=211.27.124.218;rport=12889
From: "8006" <sip:8006@my.own.domain>;tag=1b3e70a
To: <sip:8001@my.own.domain>;tag=as05ce9243
Call-ID: 671d7eb3169a29b318535e9710760c3d@0:0:0:0:0:0:0:0
CSeq: 2 INVITE
Server: Asterisk PBX 13.12.2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:8001@137.147.146.218:5060>
Content-Type: application/sdp
Content-Length: 353

v=0
o=root 1399188747 1399188747 IN IP4 137.147.146.218
s=Asterisk PBX 13.12.2
c=IN IP4 137.147.146.218
b=CT:384
t=0 0
m=audio 13874 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 17168 RTP/AVP 99
a=rtpmap:99 H264/90000
a=fmtp:99 profile-level-id=4DE01F
a=sendrecv

---
       > 0x76308208 -- Probation passed - setting RTP source address to 192.168.1.110:20001
       > 0x763052c8 -- Probation passed - setting RTP source address to 192.168.1.110:20000

<--- SIP read from UDP:211.27.124.218:12889 --->
ACK sip:8001@137.147.146.218:5060 SIP/2.0
Call-ID: 671d7eb3169a29b318535e9710760c3d@0:0:0:0:0:0:0:0
CSeq: 2 ACK
Via: SIP/2.0/UDP 211.27.124.218:12889;branch=z9hG4bK-333636-51d2e43ec7bed9a921f5038f7ef2dd84
From: "8006" <sip:8006@my.own.domain>;tag=1b3e70a
To: <sip:8001@my.own.domain>;tag=as05ce9243
Max-Forwards: 70
Authorization: Digest username="8006",realm="asterisk",nonce="3cb4a102",uri="sip:8001@my.own.domain",response="6d463a91bda7c584499c21ff6a45a2b2",algorithm=MD5
Contact: "8006" <sip:8006@211.27.124.218:12889;transport=udp;registering_acc=my_own_domain>
User-Agent: Jitsi2.8.5426Windows 7
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---

<--- SIP read from UDP:211.27.124.218:12889 --->
ACK sip:8001@137.147.146.218:5060 SIP/2.0
Call-ID: 671d7eb3169a29b318535e9710760c3d@0:0:0:0:0:0:0:0
CSeq: 2 ACK
Via: SIP/2.0/UDP 211.27.124.218:12889;branch=z9hG4bK-333636-51d2e43ec7bed9a921f5038f7ef2dd84
From: "8006" <sip:8006@my.own.domain>;tag=1b3e70a
To: <sip:8001@my.own.domain>;tag=as05ce9243
Max-Forwards: 70
Authorization: Digest username="8006",realm="asterisk",nonce="3cb4a102",uri="sip:8001@my.own.domain",response="6d463a91bda7c584499c21ff6a45a2b2",algorithm=MD5
Contact: "8006" <sip:8006@211.27.124.218:12889;transport=udp;registering_acc=my_own_domain>
User-Agent: Jitsi2.8.5426Windows 7
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
Reliably Transmitting (NAT) to 192.168.1.110:5060:
INFO sip:8001@192.168.1.110:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.250:5060;branch=z9hG4bK3a567065;rport
Max-Forwards: 70
From: "8006" <sip:8006@192.168.1.250>;tag=as5d9d25c4
To: <sip:8001@192.168.1.110:5060>;tag=1227522572
Contact: <sip:8006@192.168.1.250:5060>
Call-ID: 20660e4f2971b5a3318c8aa24206e320@192.168.1.250:5060
CSeq: 103 INFO
User-Agent: Asterisk PBX 13.12.2
Content-Type: application/media_control+xml
Content-Length: 205

<?xml version="1.0" encoding="utf-8" ?>
 <media_control>
  <vc_primitive>
   <to_encoder>
    <picture_fast_update>
    </picture_fast_update>
   </to_encoder>
  </vc_primitive>
 </media_control>

---

<--- SIP read from UDP:192.168.1.110:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.250:5060;branch=z9hG4bK3a567065;rport=5060
From: "8006" <sip:8006@192.168.1.250>;tag=as5d9d25c4
To: <sip:8001@192.168.1.110:5060>;tag=1227522572
Call-ID: 20660e4f2971b5a3318c8aa24206e320@192.168.1.250:5060
CSeq: 103 INFO
Contact: <sip:8001@192.168.1.110:5060>
User-Agent: Dahua UAC/3.0 VTO2000A V1.200.1000.0
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog 'cf8f31b8f5bc6580d67a3b965df78ea0@0:0:0:0:0:0:0:0' in 32000 ms (Method: OPTIONS)
Reliably Transmitting (NAT) to 192.168.1.110:5060:
INFO sip:8001@192.168.1.110:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.250:5060;branch=z9hG4bK375c5e1c;rport
Max-Forwards: 70
From: "8006" <sip:8006@192.168.1.250>;tag=as5d9d25c4
To: <sip:8001@192.168.1.110:5060>;tag=1227522572
Contact: <sip:8006@192.168.1.250:5060>
Call-ID: 20660e4f2971b5a3318c8aa24206e320@192.168.1.250:5060
CSeq: 104 INFO
User-Agent: Asterisk PBX 13.12.2
Content-Type: application/media_control+xml
Content-Length: 205

<?xml version="1.0" encoding="utf-8" ?>
 <media_control>
  <vc_primitive>
   <to_encoder>
    <picture_fast_update>
    </picture_fast_update>
   </to_encoder>
  </vc_primitive>
 </media_control>

---

<--- SIP read from UDP:192.168.1.110:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.250:5060;branch=z9hG4bK375c5e1c;rport=5060
From: "8006" <sip:8006@192.168.1.250>;tag=as5d9d25c4
To: <sip:8001@192.168.1.110:5060>;tag=1227522572
Call-ID: 20660e4f2971b5a3318c8aa24206e320@192.168.1.250:5060
CSeq: 104 INFO
Contact: <sip:8001@192.168.1.110:5060>
User-Agent: Dahua UAC/3.0 VTO2000A V1.200.1000.0
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
Reliably Transmitting (NAT) to 192.168.1.110:5060:
INFO sip:8001@192.168.1.110:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.250:5060;branch=z9hG4bK71963d2a;rport
Max-Forwards: 70
From: "8006" <sip:8006@192.168.1.250>;tag=as5d9d25c4
To: <sip:8001@192.168.1.110:5060>;tag=1227522572
Contact: <sip:8006@192.168.1.250:5060>
Call-ID: 20660e4f2971b5a3318c8aa24206e320@192.168.1.250:5060
CSeq: 105 INFO
User-Agent: Asterisk PBX 13.12.2
Content-Type: application/media_control+xml
Content-Length: 205

<?xml version="1.0" encoding="utf-8" ?>
 <media_control>
  <vc_primitive>
   <to_encoder>
    <picture_fast_update>
    </picture_fast_update>
   </to_encoder>
  </vc_primitive>
 </media_control>

---

<--- SIP read from UDP:192.168.1.110:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.250:5060;branch=z9hG4bK71963d2a;rport=5060
From: "8006" <sip:8006@192.168.1.250>;tag=as5d9d25c4
To: <sip:8001@192.168.1.110:5060>;tag=1227522572
Call-ID: 20660e4f2971b5a3318c8aa24206e320@192.168.1.250:5060
CSeq: 105 INFO
Contact: <sip:8001@192.168.1.110:5060>
User-Agent: Dahua UAC/3.0 VTO2000A V1.200.1000.0
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
Reliably Transmitting (NAT) to 192.168.1.110:5060:
INFO sip:8001@192.168.1.110:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.250:5060;branch=z9hG4bK23138e56;rport
Max-Forwards: 70
From: "8006" <sip:8006@192.168.1.250>;tag=as5d9d25c4
To: <sip:8001@192.168.1.110:5060>;tag=1227522572
Contact: <sip:8006@192.168.1.250:5060>
Call-ID: 20660e4f2971b5a3318c8aa24206e320@192.168.1.250:5060
CSeq: 106 INFO
User-Agent: Asterisk PBX 13.12.2
Content-Type: application/media_control+xml
Content-Length: 205

<?xml version="1.0" encoding="utf-8" ?>
 <media_control>
  <vc_primitive>
   <to_encoder>
    <picture_fast_update>
    </picture_fast_update>
   </to_encoder>
  </vc_primitive>
 </media_control>

---

<--- SIP read from UDP:192.168.1.110:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.250:5060;branch=z9hG4bK23138e56;rport=5060
From: "8006" <sip:8006@192.168.1.250>;tag=as5d9d25c4
To: <sip:8001@192.168.1.110:5060>;tag=1227522572
Call-ID: 20660e4f2971b5a3318c8aa24206e320@192.168.1.250:5060
CSeq: 106 INFO
Contact: <sip:8001@192.168.1.110:5060>
User-Agent: Dahua UAC/3.0 VTO2000A V1.200.1000.0
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---

<--- SIP read from UDP:211.27.124.218:12889 --->
OPTIONS sip:my.own.domain SIP/2.0
Call-ID: 512668a64b6797919981686d52bae29f@0:0:0:0:0:0:0:0
CSeq: 10 OPTIONS
From: "8006" <sip:8006@my.own.domain>;tag=3ec6b7b9
To: "8006" <sip:8006@my.own.domain>
Via: SIP/2.0/UDP 211.27.124.218:12889;branch=z9hG4bK-333636-8467ff8ba3beadc57741418f1cb13202
Max-Forwards: 70
Contact: "8006" <sip:8006@211.27.124.218:12889;transport=udp;registering_acc=my_own_domain>
User-Agent: Jitsi2.8.5426Windows 7
Allow: INFO,UPDATE,OPTIONS,MESSAGE,BYE,REFER,SUBSCRIBE,ACK,CANCEL,PUBLISH,NOTIFY,INVITE
Allow-Events: refer,conference,remote-control,presence,presence.winfo,message-summary
Content-Length: 0

<------------->
--- (12 headers 0 lines) ---
Sending to 211.27.124.218:12889 (NAT)
Looking for s in default (domain my.own.domain)

<--- Transmitting (NAT) to 211.27.124.218:12889 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 211.27.124.218:12889;branch=z9hG4bK-333636-8467ff8ba3beadc57741418f1cb13202;received=211.27.124.218;rport=12889
From: "8006" <sip:8006@my.own.domain>;tag=3ec6b7b9
To: "8006" <sip:8006@my.own.domain>;tag=as6a9cbdef
Call-ID: 512668a64b6797919981686d52bae29f@0:0:0:0:0:0:0:0
CSeq: 10 OPTIONS
Server: Asterisk PBX 13.12.2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:137.147.146.218:5060>
Accept: application/sdp
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '512668a64b6797919981686d52bae29f@0:0:0:0:0:0:0:0' in 32000 ms (Method: OPTIONS)
Reliably Transmitting (NAT) to 192.168.1.110:5060:
INFO sip:8001@192.168.1.110:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.250:5060;branch=z9hG4bK620dff00;rport
Max-Forwards: 70
From: "8006" <sip:8006@192.168.1.250>;tag=as5d9d25c4
To: <sip:8001@192.168.1.110:5060>;tag=1227522572
Contact: <sip:8006@192.168.1.250:5060>
Call-ID: 20660e4f2971b5a3318c8aa24206e320@192.168.1.250:5060
CSeq: 107 INFO
User-Agent: Asterisk PBX 13.12.2
Content-Type: application/media_control+xml
Content-Length: 205

<?xml version="1.0" encoding="utf-8" ?>
 <media_control>
  <vc_primitive>
   <to_encoder>
    <picture_fast_update>
    </picture_fast_update>
   </to_encoder>
  </vc_primitive>
 </media_control>

---

<--- SIP read from UDP:192.168.1.110:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.250:5060;branch=z9hG4bK620dff00;rport=5060
From: "8006" <sip:8006@192.168.1.250>;tag=as5d9d25c4
To: <sip:8001@192.168.1.110:5060>;tag=1227522572
Call-ID: 20660e4f2971b5a3318c8aa24206e320@192.168.1.250:5060
CSeq: 107 INFO
Contact: <sip:8001@192.168.1.110:5060>
User-Agent: Dahua UAC/3.0 VTO2000A V1.200.1000.0
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---

<--- SIP read from UDP:211.27.124.218:12889 --->
OPTIONS sip:my.own.domain SIP/2.0
Call-ID: 512668a64b6797919981686d52bae29f@0:0:0:0:0:0:0:0
CSeq: 10 OPTIONS
From: "8006" <sip:8006@my.own.domain>;tag=3ec6b7b9
To: "8006" <sip:8006@my.own.domain>
Via: SIP/2.0/UDP 211.27.124.218:12889;branch=z9hG4bK-333636-8467ff8ba3beadc57741418f1cb13202
Max-Forwards: 70
Contact: "8006" <sip:8006@211.27.124.218:12889;transport=udp;registering_acc=my_own_domain>
User-Agent: Jitsi2.8.5426Windows 7
Allow: INFO,UPDATE,OPTIONS,MESSAGE,BYE,REFER,SUBSCRIBE,ACK,CANCEL,PUBLISH,NOTIFY,INVITE
Allow-Events: refer,conference,remote-control,presence,presence.winfo,message-summary
Content-Length: 0

<------------->
--- (12 headers 0 lines) ---
Looking for s in default (domain my.own.domain)

<--- Transmitting (NAT) to 211.27.124.218:12889 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 211.27.124.218:12889;branch=z9hG4bK-333636-8467ff8ba3beadc57741418f1cb13202;received=211.27.124.218;rport=12889
From: "8006" <sip:8006@my.own.domain>;tag=3ec6b7b9
To: "8006" <sip:8006@my.own.domain>;tag=as6a9cbdef
Call-ID: 512668a64b6797919981686d52bae29f@0:0:0:0:0:0:0:0
CSeq: 10 OPTIONS
Server: Asterisk PBX 13.12.2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:137.147.146.218:5060>
Accept: application/sdp
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '512668a64b6797919981686d52bae29f@0:0:0:0:0:0:0:0' in 32000 ms (Method: OPTIONS)
Reliably Transmitting (NAT) to 192.168.1.110:5060:
INFO sip:8001@192.168.1.110:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.250:5060;branch=z9hG4bK7eab0fd4;rport
Max-Forwards: 70
From: "8006" <sip:8006@192.168.1.250>;tag=as5d9d25c4
To: <sip:8001@192.168.1.110:5060>;tag=1227522572
Contact: <sip:8006@192.168.1.250:5060>
Call-ID: 20660e4f2971b5a3318c8aa24206e320@192.168.1.250:5060
CSeq: 108 INFO
User-Agent: Asterisk PBX 13.12.2
Content-Type: application/media_control+xml
Content-Length: 205

<?xml version="1.0" encoding="utf-8" ?>
 <media_control>
  <vc_primitive>
   <to_encoder>
    <picture_fast_update>
    </picture_fast_update>
   </to_encoder>
  </vc_primitive>
 </media_control>

---

<--- SIP read from UDP:192.168.1.110:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.250:5060;branch=z9hG4bK7eab0fd4;rport=5060
From: "8006" <sip:8006@192.168.1.250>;tag=as5d9d25c4
To: <sip:8001@192.168.1.110:5060>;tag=1227522572
Call-ID: 20660e4f2971b5a3318c8aa24206e320@192.168.1.250:5060
CSeq: 108 INFO
Contact: <sip:8001@192.168.1.110:5060>
User-Agent: Dahua UAC/3.0 VTO2000A V1.200.1000.0
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
Reliably Transmitting (NAT) to 192.168.1.110:5060:
INFO sip:8001@192.168.1.110:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.250:5060;branch=z9hG4bK1cf09651;rport
Max-Forwards: 70
From: "8006" <sip:8006@192.168.1.250>;tag=as5d9d25c4
To: <sip:8001@192.168.1.110:5060>;tag=1227522572
Contact: <sip:8006@192.168.1.250:5060>
Call-ID: 20660e4f2971b5a3318c8aa24206e320@192.168.1.250:5060
CSeq: 109 INFO
User-Agent: Asterisk PBX 13.12.2
Content-Type: application/media_control+xml
Content-Length: 205

<?xml version="1.0" encoding="utf-8" ?>
 <media_control>
  <vc_primitive>
   <to_encoder>
    <picture_fast_update>
    </picture_fast_update>
   </to_encoder>
  </vc_primitive>
 </media_control>
2

Is the NAT router set to port forward the video port?

If no port forwarding is set for media, the router is probably automatically adding a rule for audio, but ignoring the video.

4

I already have port forwarding on the router. It forwards TCP and UDP ports 10000 through to 22000 to the Asterisk device.

5

Below is what my extensions.conf looks like. Not sure how to explicitly allow a device make external calls? I have just named the context ‘internal’ although it actually represents all of my devices.

[internal]
exten => 8001,1,Dial(SIP/8001)

exten => 8002,1,Dial(SIP/8002)

exten => 8003,1,Dial(SIP/8003)

exten => 8004,1,Dial(SIP/8004)

exten => 8005,1,Dial(SIP/8005)

exten => 8006,1,Dial(SIP/8006)

exten => 8007,1,Dial(SIP/8007)

exten => 8008,1,Dial(SIP/8008)

exten => 8009,1,Dial(SIP/8009)

[default]
exten => s,1,Hangup
6

The SIP shows a successful establishment of both video and audio, so the problem is likely to be in the the router. You should use RTP debug, or tcpdump/wireshark, to confirm whether the video is actually getting through Asterisk.