video answer

localhost*CLI>
Audio is at 12370
Video is at 207.115.87.185:15172
Adding codec g729 to SDP
Adding codec ulaw to SDP
Adding video codec h264 to SDP
Adding video codec mpeg4 to SDP
Adding video codec vp8 to SDP
Adding video codec h263 to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 96.81.150.137:63999:
INVITE sip:9998@96.81.150.137:63999;rinstance=00e80f1c7313e909;transport=tls SIP/2.0
Via: SIP/2.0/TLS 207.115.87.185:5061;branch=z9hG4bK29dba455;rport
Max-Forwards: 70
From: "Test 1" <sip:9999@207.115.87.185>;tag=as70135efa
To: <sip:9998@96.81.150.137:63999;rinstance=00e80f1c7313e909;transport=tls>
Contact: <sip:9999@207.115.87.185:5061;transport=TLS>
Call-ID: 1c3501b8081f99441d0275e01672408c@207.115.87.185:5061
CSeq: 102 INVITE
User-Agent: Asterisk PBX 13.9.1
Date: Wed, 28 Sep 2016 19:32:45 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
P-Asserted-Identity: "Test 1" <sip:9999@207.115.87.185>
Content-Type: application/sdp
Content-Length: 638

v=0
o=root 636467806 636467806 IN IP4 207.115.87.185
s=Asterisk PBX 13.9.1
c=IN IP4 207.115.87.185
b=CT:384
t=0 0
m=audio 12370 RTP/SAVP 18 0 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:150
a=sendrecv
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:MrJGiqCXvcU7l7r4ynM0mUn87Cm1EUjggxlIdmni
m=video 15172 RTP/SAVP 99 104 100 34
a=rtpmap:99 H264/90000
a=rtpmap:104 MP4V-ES/90000
a=rtpmap:100 VP8/90000
a=rtcp-fb:* ccm fir
a=rtpmap:34 H263/90000
a=sendrecv
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:nwpgjor/yYLEB/5ZFQw0XG4ebRUTIJblbtVG/umM

---

<--- SIP read from TLS:96.81.150.137:63999 --->
SIP/2.0 100 Trying
Via: SIP/2.0/TLS 207.115.87.185:5061;branch=z9hG4bK29dba455;rport=5061
To: <sip:9998@96.81.150.137:63999;rinstance=00e80f1c7313e909;transport=tls>
From: "Test 1" <sip:9999@207.115.87.185>;tag=as70135efa
Call-ID: 1c3501b8081f99441d0275e01672408c@207.115.87.185:5061
CSeq: 102 INVITE
Content-Length: 0

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

<--- SIP read from TLS:96.81.150.137:63999 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/TLS 207.115.87.185:5061;branch=z9hG4bK29dba455;rport=5061
Contact: <sip:9998@96.81.150.137:63999;rinstance=00e80f1c7313e909;transport=tls>
To: "Test 2"<sip:9998@96.81.150.137:63999;rinstance=00e80f1c7313e909;transport=tls>;tag=25c39e1c
From: "Test 1" <sip:9999@207.115.87.185>;tag=as70135efa
Call-ID: 1c3501b8081f99441d0275e01672408c@207.115.87.185:5061
CSeq: 102 INVITE
User-Agent: Bria Stretto iOS release 3.7.2002 stamp 35780.35824
Allow-Events: talk, hold
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
sip_route_dump: route/path hop: <sip:9998@96.81.150.137:63999;rinstance=00e80f1c7313e909;transport=tls>

<--- SIP read from TLS:96.81.150.137:63999 --->
SIP/2.0 200 OK
Via: SIP/2.0/TLS 207.115.87.185:5061;branch=z9hG4bK29dba455;rport=5061
Contact: <sip:9998@96.81.150.137:63999;rinstance=00e80f1c7313e909;transport=tls>
To: <sip:9998@96.81.150.137:63999;rinstance=00e80f1c7313e909;transport=tls>;tag=25c39e1c
From: "Test 1" <sip:9999@207.115.87.185>;tag=as70135efa
Call-ID: 1c3501b8081f99441d0275e01672408c@207.115.87.185:5061
CSeq: 102 INVITE
Allow: INVITE, ACK, CANCEL, BYE, REFER, INFO, NOTIFY, OPTIONS, UPDATE, PRACK, MESSAGE, SUBSCRIBE
Content-Type: application/sdp
Supported: replaces
User-Agent: Bria Stretto iOS release 3.7.2002 stamp 35780.35824
Content-Length: 566

v=0
o=- 1831524553 3 IN IP4 10.144.60.36
s=Cpc session
c=IN IP4 10.144.60.36
t=0 0
m=audio 64062 RTP/SAVP 18 0 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:8g72twxNXKJPYGc8fnNXgnhwP1EwtMOH+XtoX1w6
a=sendrecv
m=video 51812 RTP/SAVP 99 100
a=rtpmap:99 H264/90000
a=fmtp:99 profile-level-id=42800c;packetization-mode=0
a=rtpmap:100 VP8/90000
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:8g72twxNXKJPYGc8fnNXgnhwP1EwtMOH+XtoX1w6
a=sendrecv
a=rtcp-fb:* nack pli
<------------->
--- (12 headers 19 lines) ---
Found RTP audio format 18
Found RTP audio format 0
Found RTP audio format 101
Found audio description format G729 for ID 18
Found audio description format telephone-event for ID 101
Found RTP video format 99
Found RTP video format 100
Found video description format H264 for ID 99
Found video description format VP8 for ID 100
Capabilities: us - (g729|ulaw|h264|mpeg4|vp8|h263), peer - audio=(ulaw|g729)/video=(h264|vp8)/text=(nothing), combined - (g729|ulaw|h264|vp8)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 10.144.60.36:64062
Peer video RTP is at port 10.144.60.36:51812
sip_route_dump: route/path hop: <sip:9998@96.81.150.137:63999;rinstance=00e80f1c7313e909;transport=tls>
Transmitting (NAT) to 96.81.150.137:63999:
ACK sip:9998@96.81.150.137:63999;rinstance=00e80f1c7313e909;transport=tls SIP/2.0
Via: SIP/2.0/TLS 207.115.87.185:5061;branch=z9hG4bK5b8e3d88;rport
Max-Forwards: 70
From: "Test 1" <sip:9999@207.115.87.185>;tag=as70135efa
To: <sip:9998@96.81.150.137:63999;rinstance=00e80f1c7313e909;transport=tls>;tag=25c39e1c
Contact: <sip:9999@207.115.87.185:5061;transport=TLS>
Call-ID: 1c3501b8081f99441d0275e01672408c@207.115.87.185:5061
CSeq: 102 ACK
User-Agent: Asterisk PBX 13.9.1
Content-Length: 0