Video through NAT

seer_m · November 12, 2014, 2:45pm

On the quest to build a working video conferencing solution, I’ve run into a new issue. We recently opened our Asterisk 11 server to the internet and are encountering some problems. Our endpoints are a Cisco SX20 set and 2 each of Blink Qt and MicroSIP softphones, one local, the other offsite.

Register and Subscribe packets come through without issues, but Invites and Options usually get stuck somewhere, with endless retransmission attempts. However, once in a blue moon, the Invite reaches the other end and the call is established. It works this way on the local net, as well as with offsite calls, as long as the endpoints are the same or between the SX20 and MicroSIP. Calls between Blink and the SX20, even when they connect, get no media at all.

Asterisk is currently set as a mid-way stop for the RTP stream (directmedia=no and directrtpsetup=no, although as I understand it, this is standard practice behind a NAT), to avoid having to open the SX20 to the Wild Wild Web. I’ve defined local and external addresses, a bindport and set “nat=yes” because using “comedia,force_rport” got me absolutely nowhere. According to our firewall gatekeeper, all UDP traffic is forwarded to Asterisk and the RTP port range is open.

seer_m · November 13, 2014, 12:46pm

EDIT: Wrong logs.

EDIT2: A little progress update. Offsite calls work reliably, local calls tend to fail. Tried the same call twice without changing the configuration, one of them passed, the other didn’t ring. I’ve been scratching my head at the logs for a good while now, but still haven’t seen where things went wrong.

The call that passed:

<--- SIP read from UDP:|Firewall IP|:6978 --->
INVITE sip:End-2@|Asterisk IP| SIP/2.0
Via: SIP/2.0/UDP |End-1 IP|:5060;branch=z9hG4bK96be2665dcfb8ce47ea2564629dc46ee.1;rport
Call-ID: 92fe43e827c71e454085072249abc45b
CSeq: 100 INVITE
Contact: <sip:End-1@|End-1 IP|:5060>
From: "Jiggy" <sip:End-1@|Asterisk IP|>;tag=67f7e50bae3091d0
To: <sip:End-2@|Asterisk IP|>
Max-Forwards: 70
Route: <sip:|Asterisk IP|;lr>
Allow: INVITE,ACK,CANCEL,BYE,UPDATE,INFO,OPTIONS,REFER,NOTIFY
User-Agent: TANDBERG/518 (TC6.3.0.3d8e7d1)
Supported: replaces,100rel,timer,gruu,path,outbound
Session-Expires: 1800
Content-Type: application/sdp
Content-Length: 2439

v=0
o=tandberg 88 3 IN IP4 |End-1 IP|
s=-
c=IN IP4 |End-1 IP|
b=AS:2048
t=0 0
m=audio 2366 RTP/AVP 107 108 104 105 9 18 8 0 101
b=TIAS:128000
a=rtpmap:107 MP4A-LATM/90000
a=fmtp:107 profile-level-id=25;object=23;bitrate=128000
a=rtpmap:108 MP4A-LATM/90000
a=fmtp:108 profile-level-id=24;object=23;bitrate=64000
a=rtpmap:104 G7221/16000
a=fmtp:104 bitrate=32000
a=rtpmap:105 G7221/16000
a=fmtp:105 bitrate=24000
a=rtpmap:9 G722/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
m=video 2368 RTP/AVP 97 126 96 34
b=TIAS:2048000
a=rtpmap:97 H264/90000
a=fmtp:97 packetization-mode=0;profile-level-id=428016;max-br=5000;max-mbps=490000;max-fs=8160;max-smbps=490000;max-fps=6000;max-rcmd-nalu-size=3133440
a=rtpmap:126 H264/90000
a=fmtp:126 packetization-mode=1;profile-level-id=428016;max-br=5000;max-mbps=490000;max-fs=8160;max-smbps=490000;max-fps=6000;max-rcmd-nalu-size=3133440
a=rtpmap:96 H263-1998/90000
a=fmtp:96 custom=1280,768,1;custom=1280,720,1;custom=1024,768,1;custom=1024,576,1;custom=800,600,1;cif4=1;custom=720,480,1;custom=640,480,1;custom=512,288,1;cif=1;custom=352,240,1;qcif=1;maxbr=20480
a=rtpmap:34 H263/90000
a=fmtp:34 cif4=2;cif=1;qcif=1;maxbr=20000
a=label:11
a=answer:full
a=content:main
a=rtcp-fb:* nack pli
a=rtcp-fb:* ccm fir
a=rtcp-fb:* ccm tmmbr
a=sendrecv
m=application 5074 UDP/BFCP *
a=setup:actpass
a=confid:1
a=userid:88
a=floorid:2 mstrm:12
a=floorctrl:c-s
a=connection:new
m=video 2370 RTP/AVP 97 126 96 34
b=TIAS:2048000
a=rtpmap:97 H264/90000
a=fmtp:97 packetization-mode=0;profile-level-id=428014;max-br=2500;max-mbps=245000;max-fs=8160;max-smbps=245000;max-fps=6000;max-rcmd-nalu-size=3133440
a=rtpmap:126 H264/90000
a=fmtp:126 packetization-mode=1;profile-level-id=428014;max-br=2500;max-mbps=245000;max-fs=8160;max-smbps=245000;max-fps=6000;max-rcmd-nalu-size=3133440
a=rtpmap:96 H263-1998/90000
a=fmtp:96 custom=1280,768,1;custom=1280,720,1;custom=1024,768,1;custom=1024,576,1;custom=800,600,1;cif4=1;custom=720,480,1;custom=640,480,1;custom=512,288,1;cif=1;custom=352,240,1;qcif=1;maxbr=20480
a=rtpmap:34 H263/90000
a=fmtp:34 cif4=2;cif=1;qcif=1;maxbr=20000
a=label:12
a=content:slides
a=rtcp-fb:* nack pli
a=rtcp-fb:* ccm fir
a=rtcp-fb:* ccm tmmbr
a=sendrecv
m=application 2372 RTP/AVP 100
a=rtpmap:100 H224/4800
a=sendrecv

<------------->
--- (15 headers 67 lines) ---
Sending to |Firewall IP|:6978 (NAT)
Sending to |Firewall IP|:6978 (NAT)
Using INVITE request as basis request - 92fe43e827c71e454085072249abc45b
Found peer 'End-1' for 'End-1' from |Firewall IP|:6978

<--- Reliably Transmitting (NAT) to |Firewall IP|:6978 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP |End-1 IP|:5060;branch=z9hG4bK96be2665dcfb8ce47ea2564629dc46ee.1;received=|Firewall IP|;rport=6978
From: "Jiggy" <sip:End-1@|Asterisk IP|>;tag=67f7e50bae3091d0
To: <sip:End-2@|Asterisk IP|>;tag=as49db0d2d
Call-ID: 92fe43e827c71e454085072249abc45b
CSeq: 100 INVITE
Server: Asterisk PBX 11.12.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="01da70d0"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '92fe43e827c71e454085072249abc45b' in 32000 ms (Method: INVITE)

<--- SIP read from UDP:|Firewall IP|:6978 --->
ACK sip:End-2@|Asterisk IP| SIP/2.0
Via: SIP/2.0/UDP |End-1 IP|:5060;branch=z9hG4bK96be2665dcfb8ce47ea2564629dc46ee.1;rport
Call-ID: 92fe43e827c71e454085072249abc45b
CSeq: 100 ACK
From: "Jiggy" <sip:End-1@|Asterisk IP|>;tag=67f7e50bae3091d0
To: <sip:End-2@|Asterisk IP|>;tag=as49db0d2d
Route: <sip:|Asterisk IP|;lr>
User-Agent: TANDBERG/518 (TC6.3.0.3d8e7d1)
Content-Length: 0


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

<--- SIP read from UDP:|Firewall IP|:6978 --->
INVITE sip:End-2@|Asterisk IP| SIP/2.0
Via: SIP/2.0/UDP |End-1 IP|:5060;branch=z9hG4bKd296a197740df3f6acf4d2597b71c194.1;rport
Call-ID: 92fe43e827c71e454085072249abc45b
CSeq: 101 INVITE
Contact: <sip:End-1@|End-1 IP|:5060>
From: "Jiggy" <sip:End-1@|Asterisk IP|>;tag=67f7e50bae3091d0
To: <sip:End-2@|Asterisk IP|>
Max-Forwards: 70
Route: <sip:|Asterisk IP|;lr>
Allow: INVITE,ACK,CANCEL,BYE,UPDATE,INFO,OPTIONS,REFER,NOTIFY
User-Agent: TANDBERG/518 (TC6.3.0.3d8e7d1)
Authorization: Digest nonce="01da70d0", realm="asterisk", username="End-1", uri="sip:|Asterisk IP|", response="114a6e0f638dd16fca0aa79bfe8114f0", algorithm=MD5
Supported: replaces,100rel,timer,gruu,path,outbound
Session-Expires: 1800
Content-Type: application/sdp
Content-Length: 2439

v=0
o=tandberg 88 3 IN IP4 |End-1 IP|
s=-
c=IN IP4 |End-1 IP|
b=AS:2048
t=0 0
m=audio 2366 RTP/AVP 107 108 104 105 9 18 8 0 101
b=TIAS:128000
a=rtpmap:107 MP4A-LATM/90000
a=fmtp:107 profile-level-id=25;object=23;bitrate=128000
a=rtpmap:108 MP4A-LATM/90000
a=fmtp:108 profile-level-id=24;object=23;bitrate=64000
a=rtpmap:104 G7221/16000
a=fmtp:104 bitrate=32000
a=rtpmap:105 G7221/16000
a=fmtp:105 bitrate=24000
a=rtpmap:9 G722/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
m=video 2368 RTP/AVP 97 126 96 34
b=TIAS:2048000
a=rtpmap:97 H264/90000
a=fmtp:97 packetization-mode=0;profile-level-id=428016;max-br=5000;max-mbps=490000;max-fs=8160;max-smbps=490000;max-fps=6000;max-rcmd-nalu-size=3133440
a=rtpmap:126 H264/90000
a=fmtp:126 packetization-mode=1;profile-level-id=428016;max-br=5000;max-mbps=490000;max-fs=8160;max-smbps=490000;max-fps=6000;max-rcmd-nalu-size=3133440
a=rtpmap:96 H263-1998/90000
a=fmtp:96 custom=1280,768,1;custom=1280,720,1;custom=1024,768,1;custom=1024,576,1;custom=800,600,1;cif4=1;custom=720,480,1;custom=640,480,1;custom=512,288,1;cif=1;custom=352,240,1;qcif=1;maxbr=20480
a=rtpmap:34 H263/90000
a=fmtp:34 cif4=2;cif=1;qcif=1;maxbr=20000
a=label:11
a=answer:full
a=content:main
a=rtcp-fb:* nack pli
a=rtcp-fb:* ccm fir
a=rtcp-fb:* ccm tmmbr
a=sendrecv
m=application 5074 UDP/BFCP *
a=setup:actpass
a=confid:1
a=userid:88
a=floorid:2 mstrm:12
a=floorctrl:c-s
a=connection:new
m=video 2370 RTP/AVP 97 126 96 34
b=TIAS:2048000
a=rtpmap:97 H264/90000
a=fmtp:97 packetization-mode=0;profile-level-id=428014;max-br=2500;max-mbps=245000;max-fs=8160;max-smbps=245000;max-fps=6000;max-rcmd-nalu-size=3133440
a=rtpmap:126 H264/90000
a=fmtp:126 packetization-mode=1;profile-level-id=428014;max-br=2500;max-mbps=245000;max-fs=8160;max-smbps=245000;max-fps=6000;max-rcmd-nalu-size=3133440
a=rtpmap:96 H263-1998/90000
a=fmtp:96 custom=1280,768,1;custom=1280,720,1;custom=1024,768,1;custom=1024,576,1;custom=800,600,1;cif4=1;custom=720,480,1;custom=640,480,1;custom=512,288,1;cif=1;custom=352,240,1;qcif=1;maxbr=20480
a=rtpmap:34 H263/90000
a=fmtp:34 cif4=2;cif=1;qcif=1;maxbr=20000
a=label:12
a=content:slides
a=rtcp-fb:* nack pli
a=rtcp-fb:* ccm fir
a=rtcp-fb:* ccm tmmbr
a=sendrecv
m=application 2372 RTP/AVP 100
a=rtpmap:100 H224/4800
a=sendrecv

<------------->
--- (16 headers 67 lines) ---
Sending to |Firewall IP|:6978 (NAT)
Using INVITE request as basis request - 92fe43e827c71e454085072249abc45b
Found peer 'End-1' for 'End-1' from |Firewall IP|:6978
  == Using SIP VIDEO CoS mark 6
  == Using SIP RTP CoS mark 5
Found RTP audio format 107
Found RTP audio format 108
Found RTP audio format 104
Found RTP audio format 105
Found RTP audio format 9
Found RTP audio format 18
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 101
Found unknown media description format MP4A-LATM for ID 107
Found unknown media description format MP4A-LATM for ID 108
Found audio description format G7221 for ID 104
Found audio description format G7221 for ID 105
[Nov 13 15:23:17] WARNING[3425][C-00000f2c]: chan_sip.c:11009 process_sdp_a_audio: Got Siren7 offer at 24000 bps, but only 32000 bps supported; ignoring.
Found audio description format G722 for ID 9
Found audio description format G729 for ID 18
Found audio description format PCMA for ID 8
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Found RTP video format 97
Found RTP video format 126
Found RTP video format 96
Found RTP video format 34
Found video description format H264 for ID 97
Found video description format H264 for ID 126
Found video description format H263-1998 for ID 96
Found video description format H263 for ID 34
[Nov 13 15:23:17] WARNING[3425][C-00000f2c]: chan_sip.c:10094 process_sdp: Declining non-primary video stream: video 2370 RTP/AVP 97 126 96 34
Capabilities: us - (gsm|ulaw|alaw|speex|h264), peer - audio=(ulaw|alaw|g729|g722|siren7)/video=(h263|h263p|h264)/text=(nothing), combined - (ulaw|alaw|h264)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port |End-1 IP|:2366
Peer video RTP is at port |End-1 IP|:2368
Looking for End-2 in internal (domain |Asterisk IP|)
list_route: hop: <sip:End-1@|End-1 IP|:5060>

<--- Transmitting (NAT) to |Firewall IP|:6978 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP |End-1 IP|:5060;branch=z9hG4bKd296a197740df3f6acf4d2597b71c194.1;received=|Firewall IP|;rport=6978
From: "Jiggy" <sip:End-1@|Asterisk IP|>;tag=67f7e50bae3091d0
To: <sip:End-2@|Asterisk IP|>
Call-ID: 92fe43e827c71e454085072249abc45b
CSeq: 101 INVITE
Server: Asterisk PBX 11.12.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:End-2@|Asterisk IP|:5060>
Content-Length: 0


<------------>
    -- Executing [End-2@internal:1] Dial("SIP/End-1-00000211", "SIP/End-2") in new stack
  == Using SIP VIDEO CoS mark 6
  == Using SIP RTP CoS mark 5
Audio is at 30126
Video is at |Asterisk IP|:30110
Adding codec 100003 (ulaw) to SDP
Adding video codec 200004 (h264) to SDP
Adding codec 100009 (speex) to SDP
Adding codec 100004 (alaw) to SDP
Adding codec 100002 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to |Firewall IP|:8567:
INVITE sip:End-2@|Firewall IP|:8567;ob SIP/2.0
Via: SIP/2.0/UDP |Asterisk IP|:5060;branch=z9hG4bK324330a2;rport
Max-Forwards: 70
From: "Jiggy" <sip:End-1@|Asterisk IP|>;tag=as5b8c5cfa
To: <sip:End-2@|Firewall IP|:8567;ob>
Contact: <sip:End-1@|Asterisk IP|:5060>
Call-ID: 516a78b21b2dbeaa2c6ff44637728500@|Asterisk IP|:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 11.12.0
Date: Thu, 13 Nov 2014 13:23:17 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 537

v=0
o=root 267104041 267104041 IN IP4 |Asterisk IP|
s=Asterisk PBX 11.12.0
c=IN IP4 |Asterisk IP|
b=CT:384
t=0 0
m=audio 30126 RTP/AVP 0 110 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:110 speex/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
m=video 30110 RTP/AVP 99
a=rtpmap:99 H264/90000
a=fmtp:99 profile-level-id=428016;max-mbps=490000;max-fs=8160;max-br=5000;max-smbps=490000;max-fps=6000;packetization-mode=0;max-rcmd-nalu-size=3133440
a=sendrecv

---
    -- Called SIP/End-2

<--- SIP read from UDP:|Firewall IP|:8567 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP |Asterisk IP|:5060;rport=5060;received=|Asterisk IP|;branch=z9hG4bK324330a2
Call-ID: 516a78b21b2dbeaa2c6ff44637728500@|Asterisk IP|:5060
From: "Jiggy" <sip:End-1@|Asterisk IP|>;tag=as5b8c5cfa
To: <sip:End-2@|Firewall IP|;ob>
CSeq: 102 INVITE
Content-Length:  0


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

<--- SIP read from UDP:|Firewall IP|:8567 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP |Asterisk IP|:5060;rport=5060;received=|Asterisk IP|;branch=z9hG4bK324330a2
Call-ID: 516a78b21b2dbeaa2c6ff44637728500@|Asterisk IP|:5060
From: "Jiggy" <sip:End-1@|Asterisk IP|>;tag=as5b8c5cfa
To: <sip:End-2@|Firewall IP|;ob>;tag=3e155c9ca088485c966930cec5525376
CSeq: 102 INVITE
Contact: <sip:End-2@|Firewall IP|:8567;ob>
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Content-Length:  0


<------------->
--- (9 headers 0 lines) ---
list_route: hop: <sip:End-2@|Firewall IP|:8567;ob>
    -- SIP/End-2-00000212 is ringing

<--- Transmitting (NAT) to |Firewall IP|:6978 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP |End-1 IP|:5060;branch=z9hG4bKd296a197740df3f6acf4d2597b71c194.1;received=|Firewall IP|;rport=6978
From: "Jiggy" <sip:End-1@|Asterisk IP|>;tag=67f7e50bae3091d0
To: <sip:End-2@|Asterisk IP|>;tag=as177ede74
Call-ID: 92fe43e827c71e454085072249abc45b
CSeq: 101 INVITE
Server: Asterisk PBX 11.12.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:End-2@|Asterisk IP|:5060>
Content-Length: 0


<------------>
       > 0x7fe3dc00a900 -- Probation passed - setting RTP source address to |Firewall IP|:18609
       > 0x7fe3dc00a900 -- Probation passed - setting RTP source address to |Firewall IP|:18609

<--- SIP read from UDP:|Firewall IP|:8567 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP |Asterisk IP|:5060;rport=5060;received=|Asterisk IP|;branch=z9hG4bK324330a2
Call-ID: 516a78b21b2dbeaa2c6ff44637728500@|Asterisk IP|:5060
From: "Jiggy" <sip:End-1@|Asterisk IP|>;tag=as5b8c5cfa
To: <sip:End-2@|Firewall IP|;ob>;tag=3e155c9ca088485c966930cec5525376
CSeq: 102 INVITE
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Contact: <sip:End-2@|Firewall IP|:8567;ob>
Supported: replaces, 100rel, timer, norefersub
Content-Type: application/sdp
Content-Length:   469

v=0
o=- 3624881004 3624881005 IN IP4 |End-2 IP|
s=pjmedia
b=AS:352
t=0 0
a=X-nat:0
m=audio 4008 RTP/AVP 0 101
c=IN IP4 |End-2 IP|
b=TIAS:64000
a=rtcp:4009 IN IP4 |End-2 IP|
a=sendrecv
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
m=video 4010 RTP/AVP 99
c=IN IP4 |End-2 IP|
b=TIAS:256000
a=rtcp:4011 IN IP4 |End-2 IP|
a=sendrecv
a=rtpmap:99 H264/90000
a=fmtp:99 profile-level-id=428016; packetization-mode=0

<------------->
--- (11 headers 21 lines) ---
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Found RTP video format 99
Found video description format H264 for ID 99
Capabilities: us - (gsm|ulaw|alaw|speex|h264), peer - audio=(ulaw)/video=(h264)/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 |End-2 IP|:4008
Peer video RTP is at port |End-2 IP|:4010
list_route: hop: <sip:End-2@|Firewall IP|:8567;ob>
set_destination: Parsing <sip:End-2@|Firewall IP|:8567;ob> for address/port to send to
set_destination: set destination to |Firewall IP|:8567
Transmitting (NAT) to |Firewall IP|:8567:
ACK sip:End-2@|Firewall IP|:8567;ob SIP/2.0
Via: SIP/2.0/UDP |Asterisk IP|:5060;branch=z9hG4bK6f560bd5;rport
Max-Forwards: 70
From: "Jiggy" <sip:End-1@|Asterisk IP|>;tag=as5b8c5cfa
To: <sip:End-2@|Firewall IP|:8567;ob>;tag=3e155c9ca088485c966930cec5525376
Contact: <sip:End-1@|Asterisk IP|:5060>
Call-ID: 516a78b21b2dbeaa2c6ff44637728500@|Asterisk IP|:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 11.12.0
Content-Length: 0


---
    -- SIP/End-2-00000212 answered SIP/End-1-00000211
Audio is at 30142
Video is at |Asterisk IP|:30172
Adding video codec 200004 (h264) to SDP
Adding codec 100003 (ulaw) to SDP
Adding codec 100004 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (NAT) to |Firewall IP|:6978 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP |End-1 IP|:5060;branch=z9hG4bKd296a197740df3f6acf4d2597b71c194.1;received=|Firewall IP|;rport=6978
From: "Jiggy" <sip:End-1@|Asterisk IP|>;tag=67f7e50bae3091d0
To: <sip:End-2@|Asterisk IP|>;tag=as177ede74
Call-ID: 92fe43e827c71e454085072249abc45b
CSeq: 101 INVITE
Server: Asterisk PBX 11.12.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:End-2@|Asterisk IP|:5060>
Content-Type: application/sdp
Require: timer
Content-Length: 576

v=0
o=root 1974399756 1974399756 IN IP4 |Asterisk IP|
s=Asterisk PBX 11.12.0
c=IN IP4 |Asterisk IP|
b=CT:384
t=0 0
m=audio 30142 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
m=video 30172 RTP/AVP 97
a=rtpmap:97 H264/90000
a=fmtp:97 profile-level-id=428016;max-mbps=490000;max-fs=8160;max-br=5000;max-smbps=490000;max-fps=6000;packetization-mode=0;max-rcmd-nalu-size=3133440
a=sendrecv
m=application 0 UDP/BFCP *
m=video 0 RTP/AVP 97 126 96 34
m=application 0 RTP/AVP 100

<------------>
    -- Locally bridging SIP/End-1-00000211 and SIP/End-2-00000212
       > 0x7fe3dc00a900 -- Probation passed - setting RTP source address to |Firewall IP|:18609
       > 0x7fe3dc05b7b0 -- Probation passed - setting RTP source address to |Firewall IP|:40307
       > 0x7fe3dc05b7b0 -- Probation passed - setting RTP source address to |Firewall IP|:40307
       > 0x7fe3dc00a900 -- Probation passed - setting RTP source address to |Firewall IP|:18609

<--- SIP read from UDP:|Firewall IP|:6978 --->
ACK sip:End-2@|Asterisk IP|:5060 SIP/2.0
Via: SIP/2.0/UDP |End-1 IP|:5060;branch=z9hG4bK178774196f410251d9418007f1a0d4e8.1;rport
Call-ID: 92fe43e827c71e454085072249abc45b
CSeq: 101 ACK
From: "Jiggy" <sip:End-1@|Asterisk IP|>;tag=67f7e50bae3091d0
To: <sip:End-2@|Asterisk IP|>;tag=as177ede74
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,UPDATE,INFO,OPTIONS,REFER,NOTIFY
User-Agent: TANDBERG/518 (TC6.3.0.3d8e7d1)
Authorization: Digest nonce="01da70d0", realm="asterisk", username="End-1", uri="sip:|Asterisk IP|", response="114a6e0f638dd16fca0aa79bfe8114f0", algorithm=MD5
Supported: replaces,100rel,timer,gruu,path,outbound
Content-Length: 0


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

<--- SIP read from UDP:|Firewall IP|:6978 --->
INVITE sip:End-2@|Asterisk IP|:5060 SIP/2.0
Via: SIP/2.0/UDP |End-1 IP|:5060;branch=z9hG4bKc33329dd083d88df095b5e93151d459a.1;rport
Call-ID: 92fe43e827c71e454085072249abc45b
CSeq: 102 INVITE
Contact: <sip:End-1@|End-1 IP|:5060>
From: "Jiggy" <sip:End-1@|Asterisk IP|>;tag=67f7e50bae3091d0
To: <sip:End-2@|Asterisk IP|>;tag=as177ede74
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,UPDATE,INFO,OPTIONS,REFER,NOTIFY
User-Agent: TANDBERG/518 (TC6.3.0.3d8e7d1)
Authorization: Digest nonce="01da70d0", realm="asterisk", username="End-1", uri="sip:|Asterisk IP|", response="114a6e0f638dd16fca0aa79bfe8114f0", algorithm=MD5
Supported: replaces,100rel,timer,gruu,path,outbound
Require: timer
Session-Expires: 1800;refresher=uas
Min-SE: 90
Content-Type: application/sdp
Content-Length: 2439

v=0
o=tandberg 88 4 IN IP4 |End-1 IP|
s=-
c=IN IP4 |End-1 IP|
b=AS:2048
t=0 0
m=audio 2366 RTP/AVP 107 108 104 105 9 18 8 0 101
b=TIAS:128000
a=rtpmap:107 MP4A-LATM/90000
a=fmtp:107 profile-level-id=25;object=23;bitrate=128000
a=rtpmap:108 MP4A-LATM/90000
a=fmtp:108 profile-level-id=24;object=23;bitrate=64000
a=rtpmap:104 G7221/16000
a=fmtp:104 bitrate=32000
a=rtpmap:105 G7221/16000
a=fmtp:105 bitrate=24000
a=rtpmap:9 G722/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
m=video 2368 RTP/AVP 97 126 96 34
b=TIAS:2048000
a=rtpmap:97 H264/90000
a=fmtp:97 packetization-mode=0;profile-level-id=428016;max-br=5000;max-mbps=490000;max-fs=8160;max-smbps=490000;max-fps=6000;max-rcmd-nalu-size=3133440
a=rtpmap:126 H264/90000
a=fmtp:126 packetization-mode=1;profile-level-id=428016;max-br=5000;max-mbps=490000;max-fs=8160;max-smbps=490000;max-fps=6000;max-rcmd-nalu-size=3133440
a=rtpmap:96 H263-1998/90000
a=fmtp:96 custom=1280,768,1;custom=1280,720,1;custom=1024,768,1;custom=1024,576,1;custom=800,600,1;cif4=1;custom=720,480,1;custom=640,480,1;custom=512,288,1;cif=1;custom=352,240,1;qcif=1;maxbr=20480
a=rtpmap:34 H263/90000
a=fmtp:34 cif4=2;cif=1;qcif=1;maxbr=20000
a=label:11
a=answer:full
a=content:main
a=rtcp-fb:* nack pli
a=rtcp-fb:* ccm fir
a=rtcp-fb:* ccm tmmbr
a=sendrecv
m=application 5074 TCP/BFCP *
a=setup:actpass
a=confid:1
a=userid:88
a=floorid:2 mstrm:12
a=floorctrl:c-s
a=connection:new
m=video 2370 RTP/AVP 97 126 96 34
b=TIAS:2048000
a=rtpmap:97 H264/90000
a=fmtp:97 packetization-mode=0;profile-level-id=428014;max-br=2500;max-mbps=245000;max-fs=8160;max-smbps=245000;max-fps=6000;max-rcmd-nalu-size=3133440
a=rtpmap:126 H264/90000
a=fmtp:126 packetization-mode=1;profile-level-id=428014;max-br=2500;max-mbps=245000;max-fs=8160;max-smbps=245000;max-fps=6000;max-rcmd-nalu-size=3133440
a=rtpmap:96 H263-1998/90000
a=fmtp:96 custom=1280,768,1;custom=1280,720,1;custom=1024,768,1;custom=1024,576,1;custom=800,600,1;cif4=1;custom=720,480,1;custom=640,480,1;custom=512,288,1;cif=1;custom=352,240,1;qcif=1;maxbr=20480
a=rtpmap:34 H263/90000
a=fmtp:34 cif4=2;cif=1;qcif=1;maxbr=20000
a=label:12
a=content:slides
a=rtcp-fb:* nack pli
a=rtcp-fb:* ccm fir
a=rtcp-fb:* ccm tmmbr
a=sendrecv
m=application 2372 RTP/AVP 100
a=rtpmap:100 H224/4800
a=sendrecv

<------------->
--- (17 headers 67 lines) ---
Sending to |Firewall IP|:6978 (NAT)
Found RTP audio format 107
Found RTP audio format 108
Found RTP audio format 104
Found RTP audio format 105
Found RTP audio format 9
Found RTP audio format 18
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 101
Found unknown media description format MP4A-LATM for ID 107
Found unknown media description format MP4A-LATM for ID 108
Found audio description format G7221 for ID 104
Found audio description format G7221 for ID 105
[Nov 13 15:23:26] WARNING[3425][C-00000f2c]: chan_sip.c:11009 process_sdp_a_audio: Got Siren7 offer at 24000 bps, but only 32000 bps supported; ignoring.
Found audio description format G722 for ID 9
Found audio description format G729 for ID 18
Found audio description format PCMA for ID 8
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Found RTP video format 97
Found RTP video format 126
Found RTP video format 96
Found RTP video format 34
Found video description format H264 for ID 97
Found video description format H264 for ID 126
Found video description format H263-1998 for ID 96
Found video description format H263 for ID 34
[Nov 13 15:23:26] WARNING[3425][C-00000f2c]: chan_sip.c:10094 process_sdp: Declining non-primary video stream: video 2370 RTP/AVP 97 126 96 34
Capabilities: us - (gsm|ulaw|alaw|speex|h264), peer - audio=(ulaw|alaw|g729|g722|siren7)/video=(h263|h263p|h264)/text=(nothing), combined - (ulaw|alaw|h264)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port |End-1 IP|:2366
Peer video RTP is at port |End-1 IP|:2368

<--- Transmitting (NAT) to |Firewall IP|:6978 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP |End-1 IP|:5060;branch=z9hG4bKc33329dd083d88df095b5e93151d459a.1;received=|Firewall IP|;rport=6978
From: "Jiggy" <sip:End-1@|Asterisk IP|>;tag=67f7e50bae3091d0
To: <sip:End-2@|Asterisk IP|>;tag=as177ede74
Call-ID: 92fe43e827c71e454085072249abc45b
CSeq: 102 INVITE
Server: Asterisk PBX 11.12.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:End-2@|Asterisk IP|:5060>
Content-Length: 0


<------------>
Audio is at 30142
Video is at |Asterisk IP|:30172
Adding video codec 200004 (h264) to SDP
Adding codec 100003 (ulaw) to SDP
Adding codec 100004 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (NAT) to |Firewall IP|:6978 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP |End-1 IP|:5060;branch=z9hG4bKc33329dd083d88df095b5e93151d459a.1;received=|Firewall IP|;rport=6978
From: "Jiggy" <sip:End-1@|Asterisk IP|>;tag=67f7e50bae3091d0
To: <sip:End-2@|Asterisk IP|>;tag=as177ede74
Call-ID: 92fe43e827c71e454085072249abc45b
CSeq: 102 INVITE
Server: Asterisk PBX 11.12.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:End-2@|Asterisk IP|:5060>
Content-Type: application/sdp
Require: timer
Content-Length: 576

v=0
o=root 1974399756 1974399757 IN IP4 |Asterisk IP|
s=Asterisk PBX 11.12.0
c=IN IP4 |Asterisk IP|
b=CT:384
t=0 0
m=audio 30142 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
m=video 30172 RTP/AVP 97
a=rtpmap:97 H264/90000
a=fmtp:97 profile-level-id=428016;max-mbps=490000;max-fs=8160;max-br=5000;max-smbps=490000;max-fps=6000;packetization-mode=0;max-rcmd-nalu-size=3133440
a=sendrecv
m=application 0 TCP/BFCP *
m=video 0 RTP/AVP 97 126 96 34
m=application 0 RTP/AVP 100

<------------>
       > 0x7fe408264350 -- Probation passed - setting RTP source address to |Firewall IP|:55259
       > 0x7fe408264350 -- Probation passed - setting RTP source address to |Firewall IP|:55259

<--- SIP read from UDP:|Firewall IP|:6978 --->
ACK sip:End-2@|Asterisk IP|:5060 SIP/2.0
Via: SIP/2.0/UDP |End-1 IP|:5060;branch=z9hG4bKcb9c88516bda6e68f67ee9056aa3afca.1;rport
Call-ID: 92fe43e827c71e454085072249abc45b
CSeq: 102 ACK
From: "Jiggy" <sip:End-1@|Asterisk IP|>;tag=67f7e50bae3091d0
To: <sip:End-2@|Asterisk IP|>;tag=as177ede74
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,UPDATE,INFO,OPTIONS,REFER,NOTIFY
User-Agent: TANDBERG/518 (TC6.3.0.3d8e7d1)
Authorization: Digest nonce="01da70d0", realm="asterisk", username="End-1", uri="sip:|Asterisk IP|", response="114a6e0f638dd16fca0aa79bfe8114f0", algorithm=MD5
Supported: replaces,100rel,timer,gruu,path,outbound
Content-Length: 0


<------------->
--- (12 headers 0 lines) ---
       > 0x7fe4082722f0 -- Probation passed - setting RTP source address to |Firewall IP|:34295
       > 0x7fe4082722f0 -- Probation passed - setting RTP source address to |Firewall IP|:34295

<--- SIP read from UDP:|Firewall IP|:6978 --->
INFO sip:End-2@|Asterisk IP|:5060 SIP/2.0
Via: SIP/2.0/UDP |End-1 IP|:5060;branch=z9hG4bKcb1dc3ca4714635deb0d1922ae152111.1;rport
Call-ID: 92fe43e827c71e454085072249abc45b
CSeq: 103 INFO
Contact: <sip:End-1@|End-1 IP|:5060>
From: "Jiggy" <sip:End-1@|Asterisk IP|>;tag=67f7e50bae3091d0
To: <sip:End-2@|Asterisk IP|>;tag=as177ede74
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,UPDATE,INFO,OPTIONS,REFER,NOTIFY
User-Agent: TANDBERG/518 (TC6.3.0.3d8e7d1)
Authorization: Digest nonce="01da70d0", realm="asterisk", username="End-1", uri="sip:|Asterisk IP|", response="114a6e0f638dd16fca0aa79bfe8114f0", algorithm=MD5
Supported: replaces,100rel,timer,gruu,path,outbound
Content-Type: application/media_control+xml
Content-Length: 193

<?xml version="1.0" encoding="utf-8"?><media_control><vc_primitive><to_encoder><picture_fast_update></picture_fast_update></to_encoder><stream_id>11</stream_id></vc_primitive></media_control>

<------------->
--- (14 headers 1 lines) ---
Receiving INFO!

<--- Transmitting (NAT) to |Firewall IP|:6978 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP |End-1 IP|:5060;branch=z9hG4bKcb1dc3ca4714635deb0d1922ae152111.1;received=|Firewall IP|;rport=6978
From: "Jiggy" <sip:End-1@|Asterisk IP|>;tag=67f7e50bae3091d0
To: <sip:End-2@|Asterisk IP|>;tag=as177ede74
Call-ID: 92fe43e827c71e454085072249abc45b
CSeq: 103 INFO
Server: Asterisk PBX 11.12.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


<------------>
set_destination: Parsing <sip:End-2@|Firewall IP|:8567;ob> for address/port to send to
set_destination: set destination to |Firewall IP|:8567
Reliably Transmitting (NAT) to |Firewall IP|:8567:
INFO sip:End-2@|Firewall IP|:8567;ob SIP/2.0
Via: SIP/2.0/UDP |Asterisk IP|:5060;branch=z9hG4bK5750ab10;rport
Max-Forwards: 70
From: "Jiggy" <sip:End-1@|Asterisk IP|>;tag=as5b8c5cfa
To: <sip:End-2@|Firewall IP|:8567;ob>;tag=3e155c9ca088485c966930cec5525376
Contact: <sip:End-1@|Asterisk IP|:5060>
Call-ID: 516a78b21b2dbeaa2c6ff44637728500@|Asterisk IP|:5060
CSeq: 103 INFO
User-Agent: Asterisk PBX 11.12.0
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:|Firewall IP|:8567 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP |Asterisk IP|:5060;rport=5060;received=|Asterisk IP|;branch=z9hG4bK5750ab10
Call-ID: 516a78b21b2dbeaa2c6ff44637728500@|Asterisk IP|:5060
From: "Jiggy" <sip:End-1@|Asterisk IP|>;tag=as5b8c5cfa
To: <sip:End-2@|Firewall IP|;ob>;tag=3e155c9ca088485c966930cec5525376
CSeq: 103 INFO
Content-Length:  0


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

<--- SIP read from UDP:|Firewall IP|:8567 --->
INFO sip:End-1@|Asterisk IP|:5060 SIP/2.0
Via: SIP/2.0/UDP |Firewall IP|:8567;rport;branch=z9hG4bKPjfb6c59e454b34d13b8d7b17535b31f6e
Max-Forwards: 70
From: <sip:End-2@|Firewall IP|;ob>;tag=3e155c9ca088485c966930cec5525376
To: "Jiggy" <sip:End-1@|Asterisk IP|>;tag=as5b8c5cfa
Call-ID: 516a78b21b2dbeaa2c6ff44637728500@|Asterisk IP|:5060
CSeq: 27593 INFO
User-Agent: MicroSIP/3.8.1
Content-Type: application/media_control+xml
Content-Length:   146

<?xml version="1.0" encoding="utf-8" ?><media_control><vc_primitive><to_encoder><picture_fast_update/></to_encoder></vc_primitive></media_control>
<------------->
--- (10 headers 1 lines) ---
Receiving INFO!

<--- Transmitting (NAT) to |Firewall IP|:8567 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP |Firewall IP|:8567;branch=z9hG4bKPjfb6c59e454b34d13b8d7b17535b31f6e;received=|Firewall IP|;rport=8567
From: <sip:End-2@|Firewall IP|;ob>;tag=3e155c9ca088485c966930cec5525376
To: "Jiggy" <sip:End-1@|Asterisk IP|>;tag=as5b8c5cfa
Call-ID: 516a78b21b2dbeaa2c6ff44637728500@|Asterisk IP|:5060
CSeq: 27593 INFO
Server: Asterisk PBX 11.12.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


<------------>
set_destination: Parsing <sip:End-1@|End-1 IP|:5060> for address/port to send to
set_destination: set destination to |End-1 IP|:5060
Reliably Transmitting (NAT) to |Firewall IP|:6978:
INFO sip:End-1@|End-1 IP|:5060 SIP/2.0
Via: SIP/2.0/UDP |Asterisk IP|:5060;branch=z9hG4bK4731f888;rport
Max-Forwards: 70
From: <sip:End-2@|Asterisk IP|>;tag=as177ede74
To: "Jiggy" <sip:End-1@|Asterisk IP|>;tag=67f7e50bae3091d0
Contact: <sip:End-2@|Asterisk IP|:5060>
Call-ID: 92fe43e827c71e454085072249abc45b
CSeq: 102 INFO
User-Agent: Asterisk PBX 11.12.0
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:|Firewall IP|:6978 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP |Asterisk IP|:5060;branch=z9hG4bK4731f888;received=|Asterisk IP|;rport=5060
Call-ID: 92fe43e827c71e454085072249abc45b
CSeq: 102 INFO
Contact: <sip:End-1@|End-1 IP|:5060>
From: <sip:End-2@|Asterisk IP|>;tag=as177ede74
To: "Jiggy" <sip:End-1@|Asterisk IP|>;tag=67f7e50bae3091d0
Allow: INVITE,ACK,CANCEL,BYE,UPDATE,INFO,OPTIONS,REFER,NOTIFY
Server: TANDBERG/518 (TC6.3.0.3d8e7d1)
Supported: replaces,100rel,timer,gruu,path,outbound
Content-Length: 0

The call that failed:

<--- SIP read from UDP:|Firewall IP|:56947 --->
INVITE sip:End-2@|Asterisk IP| SIP/2.0
Via: SIP/2.0/UDP |End-1 IP|:5060;branch=z9hG4bKdb6f3c5f397c9d2ba45c78b237fcb676.1;rport
Call-ID: 96df952a42f5c24edcbba7b86ab41647
CSeq: 100 INVITE
Contact: <sip:End-1@|End-1 IP|:5060>
From: "Jiggy" <sip:End-1@|Asterisk IP|>;tag=5b67d93335bdd37e
To: <sip:End-2@|Asterisk IP|>
Max-Forwards: 70
Route: <sip:|Asterisk IP|;lr>
Allow: INVITE,ACK,CANCEL,BYE,UPDATE,INFO,OPTIONS,REFER,NOTIFY
User-Agent: TANDBERG/518 (TC6.3.0.3d8e7d1)
Supported: replaces,100rel,timer,gruu,path,outbound
Session-Expires: 1800
Content-Type: application/sdp
Content-Length: 2439

v=0
o=tandberg 90 3 IN IP4 |End-1 IP|
s=-
c=IN IP4 |End-1 IP|
b=AS:2048
t=0 0
m=audio 2386 RTP/AVP 107 108 104 105 9 18 8 0 101
b=TIAS:128000
a=rtpmap:107 MP4A-LATM/90000
a=fmtp:107 profile-level-id=25;object=23;bitrate=128000
a=rtpmap:108 MP4A-LATM/90000
a=fmtp:108 profile-level-id=24;object=23;bitrate=64000
a=rtpmap:104 G7221/16000
a=fmtp:104 bitrate=32000
a=rtpmap:105 G7221/16000
a=fmtp:105 bitrate=24000
a=rtpmap:9 G722/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
m=video 2388 RTP/AVP 97 126 96 34
b=TIAS:2048000
a=rtpmap:97 H264/90000
a=fmtp:97 packetization-mode=0;profile-level-id=428016;max-br=5000;max-mbps=490000;max-fs=8160;max-smbps=490000;max-fps=6000;max-rcmd-nalu-size=3133440
a=rtpmap:126 H264/90000
a=fmtp:126 packetization-mode=1;profile-level-id=428016;max-br=5000;max-mbps=490000;max-fs=8160;max-smbps=490000;max-fps=6000;max-rcmd-nalu-size=3133440
a=rtpmap:96 H263-1998/90000
a=fmtp:96 custom=1280,768,1;custom=1280,720,1;custom=1024,768,1;custom=1024,576,1;custom=800,600,1;cif4=1;custom=720,480,1;custom=640,480,1;custom=512,288,1;cif=1;custom=352,240,1;qcif=1;maxbr=20480
a=rtpmap:34 H263/90000
a=fmtp:34 cif4=2;cif=1;qcif=1;maxbr=20000
a=label:11
a=answer:full
a=content:main
a=rtcp-fb:* nack pli
a=rtcp-fb:* ccm fir
a=rtcp-fb:* ccm tmmbr
a=sendrecv
m=application 5076 UDP/BFCP *
a=setup:actpass
a=confid:1
a=userid:90
a=floorid:2 mstrm:12
a=floorctrl:c-s
a=connection:new
m=video 2390 RTP/AVP 97 126 96 34
b=TIAS:2048000
a=rtpmap:97 H264/90000
a=fmtp:97 packetization-mode=0;profile-level-id=428014;max-br=2500;max-mbps=245000;max-fs=8160;max-smbps=245000;max-fps=6000;max-rcmd-nalu-size=3133440
a=rtpmap:126 H264/90000
a=fmtp:126 packetization-mode=1;profile-level-id=428014;max-br=2500;max-mbps=245000;max-fs=8160;max-smbps=245000;max-fps=6000;max-rcmd-nalu-size=3133440
a=rtpmap:96 H263-1998/90000
a=fmtp:96 custom=1280,768,1;custom=1280,720,1;custom=1024,768,1;custom=1024,576,1;custom=800,600,1;cif4=1;custom=720,480,1;custom=640,480,1;custom=512,288,1;cif=1;custom=352,240,1;qcif=1;maxbr=20480
a=rtpmap:34 H263/90000
a=fmtp:34 cif4=2;cif=1;qcif=1;maxbr=20000
a=label:12
a=content:slides
a=rtcp-fb:* nack pli
a=rtcp-fb:* ccm fir
a=rtcp-fb:* ccm tmmbr
a=sendrecv
m=application 2392 RTP/AVP 100
a=rtpmap:100 H224/4800
a=sendrecv

<------------->
--- (15 headers 67 lines) ---
Sending to |Firewall IP|:56947 (NAT)
Sending to |Firewall IP|:56947 (NAT)
Using INVITE request as basis request - 96df952a42f5c24edcbba7b86ab41647
Found peer 'End-1' for 'End-1' from |Firewall IP|:56947

<--- Reliably Transmitting (NAT) to |Firewall IP|:56947 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP |End-1 IP|:5060;branch=z9hG4bKdb6f3c5f397c9d2ba45c78b237fcb676.1;received=|Firewall IP|;rport=56947
From: "Jiggy" <sip:End-1@|Asterisk IP|>;tag=5b67d93335bdd37e
To: <sip:End-2@|Asterisk IP|>;tag=as320dad0f
Call-ID: 96df952a42f5c24edcbba7b86ab41647
CSeq: 100 INVITE
Server: Asterisk PBX 11.12.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="709ec2ab"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '96df952a42f5c24edcbba7b86ab41647' in 32000 ms (Method: INVITE)

<--- SIP read from UDP:|Firewall IP|:56947 --->
ACK sip:End-2@|Asterisk IP| SIP/2.0
Via: SIP/2.0/UDP |End-1 IP|:5060;branch=z9hG4bKdb6f3c5f397c9d2ba45c78b237fcb676.1;rport
Call-ID: 96df952a42f5c24edcbba7b86ab41647
CSeq: 100 ACK
From: "Jiggy" <sip:End-1@|Asterisk IP|>;tag=5b67d93335bdd37e
To: <sip:End-2@|Asterisk IP|>;tag=as320dad0f
Route: <sip:|Asterisk IP|;lr>
User-Agent: TANDBERG/518 (TC6.3.0.3d8e7d1)
Content-Length: 0


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

<--- SIP read from UDP:|Firewall IP|:56947 --->
INVITE sip:End-2@|Asterisk IP| SIP/2.0
Via: SIP/2.0/UDP |End-1 IP|:5060;branch=z9hG4bKfbd18a7b79f6f51cafe6abec02d5aea7.1;rport
Call-ID: 96df952a42f5c24edcbba7b86ab41647
CSeq: 101 INVITE
Contact: <sip:End-1@|End-1 IP|:5060>
From: "Jiggy" <sip:End-1@|Asterisk IP|>;tag=5b67d93335bdd37e
To: <sip:End-2@|Asterisk IP|>
Max-Forwards: 70
Route: <sip:|Asterisk IP|;lr>
Allow: INVITE,ACK,CANCEL,BYE,UPDATE,INFO,OPTIONS,REFER,NOTIFY
User-Agent: TANDBERG/518 (TC6.3.0.3d8e7d1)
Authorization: Digest nonce="709ec2ab", realm="asterisk", username="End-1", uri="sip:|Asterisk IP|", response="50e7991d77c04165a30c06c0e5419569", algorithm=MD5
Supported: replaces,100rel,timer,gruu,path,outbound
Session-Expires: 1800
Content-Type: application/sdp
Content-Length: 2439

v=0
o=tandberg 90 3 IN IP4 |End-1 IP|
s=-
c=IN IP4 |End-1 IP|
b=AS:2048
t=0 0
m=audio 2386 RTP/AVP 107 108 104 105 9 18 8 0 101
b=TIAS:128000
a=rtpmap:107 MP4A-LATM/90000
a=fmtp:107 profile-level-id=25;object=23;bitrate=128000
a=rtpmap:108 MP4A-LATM/90000
a=fmtp:108 profile-level-id=24;object=23;bitrate=64000
a=rtpmap:104 G7221/16000
a=fmtp:104 bitrate=32000
a=rtpmap:105 G7221/16000
a=fmtp:105 bitrate=24000
a=rtpmap:9 G722/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
m=video 2388 RTP/AVP 97 126 96 34
b=TIAS:2048000
a=rtpmap:97 H264/90000
a=fmtp:97 packetization-mode=0;profile-level-id=428016;max-br=5000;max-mbps=490000;max-fs=8160;max-smbps=490000;max-fps=6000;max-rcmd-nalu-size=3133440
a=rtpmap:126 H264/90000
a=fmtp:126 packetization-mode=1;profile-level-id=428016;max-br=5000;max-mbps=490000;max-fs=8160;max-smbps=490000;max-fps=6000;max-rcmd-nalu-size=3133440
a=rtpmap:96 H263-1998/90000
a=fmtp:96 custom=1280,768,1;custom=1280,720,1;custom=1024,768,1;custom=1024,576,1;custom=800,600,1;cif4=1;custom=720,480,1;custom=640,480,1;custom=512,288,1;cif=1;custom=352,240,1;qcif=1;maxbr=20480
a=rtpmap:34 H263/90000
a=fmtp:34 cif4=2;cif=1;qcif=1;maxbr=20000
a=label:11
a=answer:full
a=content:main
a=rtcp-fb:* nack pli
a=rtcp-fb:* ccm fir
a=rtcp-fb:* ccm tmmbr
a=sendrecv
m=application 5076 UDP/BFCP *
a=setup:actpass
a=confid:1
a=userid:90
a=floorid:2 mstrm:12
a=floorctrl:c-s
a=connection:new
m=video 2390 RTP/AVP 97 126 96 34
b=TIAS:2048000
a=rtpmap:97 H264/90000
a=fmtp:97 packetization-mode=0;profile-level-id=428014;max-br=2500;max-mbps=245000;max-fs=8160;max-smbps=245000;max-fps=6000;max-rcmd-nalu-size=3133440
a=rtpmap:126 H264/90000
a=fmtp:126 packetization-mode=1;profile-level-id=428014;max-br=2500;max-mbps=245000;max-fs=8160;max-smbps=245000;max-fps=6000;max-rcmd-nalu-size=3133440
a=rtpmap:96 H263-1998/90000
a=fmtp:96 custom=1280,768,1;custom=1280,720,1;custom=1024,768,1;custom=1024,576,1;custom=800,600,1;cif4=1;custom=720,480,1;custom=640,480,1;custom=512,288,1;cif=1;custom=352,240,1;qcif=1;maxbr=20480
a=rtpmap:34 H263/90000
a=fmtp:34 cif4=2;cif=1;qcif=1;maxbr=20000
a=label:12
a=content:slides
a=rtcp-fb:* nack pli
a=rtcp-fb:* ccm fir
a=rtcp-fb:* ccm tmmbr
a=sendrecv
m=application 2392 RTP/AVP 100
a=rtpmap:100 H224/4800
a=sendrecv

<------------->
--- (16 headers 67 lines) ---
Sending to |Firewall IP|:56947 (NAT)
Using INVITE request as basis request - 96df952a42f5c24edcbba7b86ab41647
Found peer 'End-1' for 'End-1' from |Firewall IP|:56947
  == Using SIP VIDEO CoS mark 6
  == Using SIP RTP CoS mark 5
Found RTP audio format 107
Found RTP audio format 108
Found RTP audio format 104
Found RTP audio format 105
Found RTP audio format 9
Found RTP audio format 18
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 101
Found unknown media description format MP4A-LATM for ID 107
Found unknown media description format MP4A-LATM for ID 108
Found audio description format G7221 for ID 104
Found audio description format G7221 for ID 105
[Nov 13 15:27:02] WARNING[3425][C-00000f2e]: chan_sip.c:11009 process_sdp_a_audio: Got Siren7 offer at 24000 bps, but only 32000 bps supported; ignoring.
Found audio description format G722 for ID 9
Found audio description format G729 for ID 18
Found audio description format PCMA for ID 8
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Found RTP video format 97
Found RTP video format 126
Found RTP video format 96
Found RTP video format 34
Found video description format H264 for ID 97
Found video description format H264 for ID 126
Found video description format H263-1998 for ID 96
Found video description format H263 for ID 34
[Nov 13 15:27:02] WARNING[3425][C-00000f2e]: chan_sip.c:10094 process_sdp: Declining non-primary video stream: video 2390 RTP/AVP 97 126 96 34
Capabilities: us - (gsm|ulaw|alaw|speex|h264), peer - audio=(ulaw|alaw|g729|g722|siren7)/video=(h263|h263p|h264)/text=(nothing), combined - (ulaw|alaw|h264)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port |End-1 IP|:2386
Peer video RTP is at port |End-1 IP|:2388
Looking for End-2 in internal (domain |Asterisk IP|)
list_route: hop: <sip:End-1@|End-1 IP|:5060>

<--- Transmitting (NAT) to |Firewall IP|:56947 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP |End-1 IP|:5060;branch=z9hG4bKfbd18a7b79f6f51cafe6abec02d5aea7.1;received=|Firewall IP|;rport=56947
From: "Jiggy" <sip:End-1@|Asterisk IP|>;tag=5b67d93335bdd37e
To: <sip:End-2@|Asterisk IP|>
Call-ID: 96df952a42f5c24edcbba7b86ab41647
CSeq: 101 INVITE
Server: Asterisk PBX 11.12.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:End-2@|Asterisk IP|:5060>
Content-Length: 0


<------------>
    -- Executing [End-2@internal:1] Dial("SIP/End-1-00000215", "SIP/End-2") in new stack
  == Using SIP VIDEO CoS mark 6
  == Using SIP RTP CoS mark 5
Audio is at 30158
Video is at |Asterisk IP|:30178
Adding codec 100003 (ulaw) to SDP
Adding video codec 200004 (h264) to SDP
Adding codec 100009 (speex) to SDP
Adding codec 100004 (alaw) to SDP
Adding codec 100002 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to |Firewall IP|:8567:
INVITE sip:End-2@|Firewall IP|:8567;ob SIP/2.0
Via: SIP/2.0/UDP |Asterisk IP|:5060;branch=z9hG4bK5d4cebe3;rport
Max-Forwards: 70
From: "Jiggy" <sip:End-1@|Asterisk IP|>;tag=as59542df2
To: <sip:End-2@|Firewall IP|:8567;ob>
Contact: <sip:End-1@|Asterisk IP|:5060>
Call-ID: 27d1df4a0aa61d616db4c8817d754fee@|Asterisk IP|:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 11.12.0
Date: Thu, 13 Nov 2014 13:27:02 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 539

v=0
o=root 1509771753 1509771753 IN IP4 |Asterisk IP|
s=Asterisk PBX 11.12.0
c=IN IP4 |Asterisk IP|
b=CT:384
t=0 0
m=audio 30158 RTP/AVP 0 110 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:110 speex/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
m=video 30178 RTP/AVP 99
a=rtpmap:99 H264/90000
a=fmtp:99 profile-level-id=428016;max-mbps=490000;max-fs=8160;max-br=5000;max-smbps=490000;max-fps=6000;packetization-mode=0;max-rcmd-nalu-size=3133440
a=sendrecv

---
    -- Called SIP/End-2
Retransmitting #1 (NAT) to |Firewall IP|:8567:
INVITE sip:End-2@|Firewall IP|:8567;ob SIP/2.0
Via: SIP/2.0/UDP |Asterisk IP|:5060;branch=z9hG4bK5d4cebe3;rport
Max-Forwards: 70
From: "Jiggy" <sip:End-1@|Asterisk IP|>;tag=as59542df2
To: <sip:End-2@|Firewall IP|:8567;ob>
Contact: <sip:End-1@|Asterisk IP|:5060>
Call-ID: 27d1df4a0aa61d616db4c8817d754fee@|Asterisk IP|:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 11.12.0
Date: Thu, 13 Nov 2014 13:27:02 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 539

v=0
o=root 1509771753 1509771753 IN IP4 |Asterisk IP|
s=Asterisk PBX 11.12.0
c=IN IP4 |Asterisk IP|
b=CT:384
t=0 0
m=audio 30158 RTP/AVP 0 110 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:110 speex/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
m=video 30178 RTP/AVP 99
a=rtpmap:99 H264/90000
a=fmtp:99 profile-level-id=428016;max-mbps=490000;max-fs=8160;max-br=5000;max-smbps=490000;max-fps=6000;packetization-mode=0;max-rcmd-nalu-size=3133440
a=sendrecv

---
Retransmitting #2 (NAT) to |Firewall IP|:8567:
INVITE sip:End-2@|Firewall IP|:8567;ob SIP/2.0
Via: SIP/2.0/UDP |Asterisk IP|:5060;branch=z9hG4bK5d4cebe3;rport
Max-Forwards: 70
From: "Jiggy" <sip:End-1@|Asterisk IP|>;tag=as59542df2
To: <sip:End-2@|Firewall IP|:8567;ob>
Contact: <sip:End-1@|Asterisk IP|:5060>
Call-ID: 27d1df4a0aa61d616db4c8817d754fee@|Asterisk IP|:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 11.12.0
Date: Thu, 13 Nov 2014 13:27:02 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 539

v=0
o=root 1509771753 1509771753 IN IP4 |Asterisk IP|
s=Asterisk PBX 11.12.0
c=IN IP4 |Asterisk IP|
b=CT:384
t=0 0
m=audio 30158 RTP/AVP 0 110 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:110 speex/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
m=video 30178 RTP/AVP 99
a=rtpmap:99 H264/90000
a=fmtp:99 profile-level-id=428016;max-mbps=490000;max-fs=8160;max-br=5000;max-smbps=490000;max-fps=6000;packetization-mode=0;max-rcmd-nalu-size=3133440
a=sendrecv

---
Retransmitting #3 (NAT) to |Firewall IP|:8567:
INVITE sip:End-2@|Firewall IP|:8567;ob SIP/2.0
Via: SIP/2.0/UDP |Asterisk IP|:5060;branch=z9hG4bK5d4cebe3;rport
Max-Forwards: 70
From: "Jiggy" <sip:End-1@|Asterisk IP|>;tag=as59542df2
To: <sip:End-2@|Firewall IP|:8567;ob>
Contact: <sip:End-1@|Asterisk IP|:5060>
Call-ID: 27d1df4a0aa61d616db4c8817d754fee@|Asterisk IP|:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 11.12.0
Date: Thu, 13 Nov 2014 13:27:02 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 539

v=0
o=root 1509771753 1509771753 IN IP4 |Asterisk IP|
s=Asterisk PBX 11.12.0
c=IN IP4 |Asterisk IP|
b=CT:384
t=0 0
m=audio 30158 RTP/AVP 0 110 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:110 speex/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
m=video 30178 RTP/AVP 99
a=rtpmap:99 H264/90000
a=fmtp:99 profile-level-id=428016;max-mbps=490000;max-fs=8160;max-br=5000;max-smbps=490000;max-fps=6000;packetization-mode=0;max-rcmd-nalu-size=3133440
a=sendrecv

---
Retransmitting #6 (NAT) to |Firewall IP|:8567:
INVITE sip:End-2@|Firewall IP|:8567;ob SIP/2.0
Via: SIP/2.0/UDP |Asterisk IP|:5060;branch=z9hG4bK0456e099;rport
Max-Forwards: 70
From: "Jiggy" <sip:End-1@|Asterisk IP|>;tag=as459a7e66
To: <sip:End-2@|Firewall IP|:8567;ob>
Contact: <sip:End-1@|Asterisk IP|:5060>
Call-ID: 2a9aeb2163b48ddb4e9bbd437ecbbc1f@|Asterisk IP|:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 11.12.0
Date: Thu, 13 Nov 2014 13:26:36 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 539

v=0
o=root 1664786311 1664786311 IN IP4 |Asterisk IP|
s=Asterisk PBX 11.12.0
c=IN IP4 |Asterisk IP|
b=CT:384
t=0 0
m=audio 30146 RTP/AVP 0 110 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:110 speex/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
m=video 30176 RTP/AVP 99
a=rtpmap:99 H264/90000
a=fmtp:99 profile-level-id=428016;max-mbps=490000;max-fs=8160;max-br=5000;max-smbps=490000;max-fps=6000;packetization-mode=0;max-rcmd-nalu-size=3133440
a=sendrecv

---
[Nov 13 15:27:08] WARNING[3425]: chan_sip.c:4024 retrans_pkt: Retransmission timeout reached on transmission 2a9aeb2163b48ddb4e9bbd437ecbbc1f@|Asterisk IP|:5060 for seqno 102 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 31999ms with no response
Really destroying SIP dialog '2a9aeb2163b48ddb4e9bbd437ecbbc1f@|Asterisk IP|:5060' Method: INVITE
Retransmitting #4 (NAT) to |Firewall IP|:8567:
INVITE sip:End-2@|Firewall IP|:8567;ob SIP/2.0
Via: SIP/2.0/UDP |Asterisk IP|:5060;branch=z9hG4bK5d4cebe3;rport
Max-Forwards: 70
From: "Jiggy" <sip:End-1@|Asterisk IP|>;tag=as59542df2
To: <sip:End-2@|Firewall IP|:8567;ob>
Contact: <sip:End-1@|Asterisk IP|:5060>
Call-ID: 27d1df4a0aa61d616db4c8817d754fee@|Asterisk IP|:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 11.12.0
Date: Thu, 13 Nov 2014 13:27:02 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 539

v=0
o=root 1509771753 1509771753 IN IP4 |Asterisk IP|
s=Asterisk PBX 11.12.0
c=IN IP4 |Asterisk IP|
b=CT:384
t=0 0
m=audio 30158 RTP/AVP 0 110 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:110 speex/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
m=video 30178 RTP/AVP 99
a=rtpmap:99 H264/90000
a=fmtp:99 profile-level-id=428016;max-mbps=490000;max-fs=8160;max-br=5000;max-smbps=490000;max-fps=6000;packetization-mode=0;max-rcmd-nalu-size=3133440
a=sendrecv