No video from Asterisk endpoint to CUCM ivr via sip trunk

david551 · October 4, 2025, 4:47pm

PJSIP Logging enabled

<--- Received SIP request (374 bytes) from UDP:172.17.67.14:5060 --->
OPTIONS sip:172.16.48.3:5060 SIP/2.0
Via: SIP/2.0/UDP 172.17.67.14:5060;branch=z9hG4bK43f75a42a800
From: <sip:172.17.67.14>;tag=505072884
To: <sip:172.16.48.3>
Date: Sat, 04 Oct 2025 12:22:28 GMT
Call-ID: c9755f80-1f11bf1f-3e89-e4311ac@172.17.67.14
User-Agent: Cisco-CUCM12.5
CSeq: 101 OPTIONS
Contact: <sip:172.17.67.14:5060>
Max-Forwards: 0
Content-Length: 0
<--- Transmitting SIP response (838 bytes) to UDP:172.17.67.14:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.17.67.14:5060;rport=5060;received=172.17.67.14;branch=z9hG4bK43f75a42a800
Call-ID: c9755f80-1f11bf1f-3e89-e4311ac@172.17.67.14
From: <sip:172.17.67.14>;tag=505072884
To: <sip:172.16.48.3>;tag=z9hG4bK43f75a42a800
CSeq: 101 OPTIONS
Accept: application/sdp, application/pidf+xml, application/dialog-info+xml, application/xpidf+xml, application/cpim-pidf+xml, application/simple-message-summary, application/simple-message-summary, application/pidf+xml, application/dialog-info+xml, message/sipfrag;version=2.0
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, INFO, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub
Accept-Encoding: identity
Accept-Language: en
Server: Asterisk PBX 22.2.0~dfsg+~cs6.15.60671435-2
Content-Length: 0
<--- Received SIP request (1414 bytes) from UDP:172.16.48.5:49517 --->
INVITE sip:7100@webcall.collnetwork.net SIP/2.0
Via: SIP/2.0/UDP 172.16.48.5:49517;rport;branch=z9hG4bKPj6ba04f9e81d5480696f0358c8a6ca70d
Max-Forwards: 70
From: <sip:1605@webcall.collnetwork.net>;tag=d80dfdedf439490ba3ec7ecde07f8de4
To: <sip:7100@webcall.collnetwork.net>
Contact: <sip:1605@172.16.48.5:49517;ob>
Call-ID: 2795b75be8dc436a82b9c72a4a65dad0
CSeq: 31340 INVITE
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 1800
Min-SE: 90
User-Agent: MicroSIP/3.22.0
Content-Type: application/sdp
Content-Length: 766

v=0
o=- 3968572955 3968572955 IN IP4 172.16.48.5
s=pjmedia
b=AS:352
t=0 0
a=X-nat:0
m=audio 4000 RTP/AVP 8 0 101
c=IN IP4 172.16.48.5
b=TIAS:64000
a=rtcp:4001 IN IP4 172.16.48.5
a=sendrecv
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ssrc:351082781 cname:3ea53a1e0f3760d1
m=video 4002 RTP/AVP 99 98 100 101
c=IN IP4 172.16.48.5
b=TIAS:256000
a=rtcp:4003 IN IP4 172.16.48.5
a=sendrecv
a=rtpmap:99 H264/90000
a=fmtp:99 profile-level-id=42e01e; packetization-mode=1
a=rtpmap:98 H263-1998/90000
a=fmtp:98 CIF=1;QCIF=1
a=rtpmap:100 VP8/90000
a=fmtp:100 max-fr=30; max-fs=580
a=rtpmap:101 VP9/90000
a=fmtp:101 max-fr=30; max-fs=580
a=ssrc:45757132 cname:3ea53a1e0f3760d1
a=rtcp-fb:* nack pli
<--- Transmitting SIP response (593 bytes) to UDP:172.16.48.5:49517 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.16.48.5:49517;rport=49517;received=172.16.48.5;branch=z9hG4bKPj6ba04f9e81d5480696f0358c8a6ca70d
Call-ID: 2795b75be8dc436a82b9c72a4a65dad0
From: <sip:1605@webcall.collnetwork.net>;tag=d80dfdedf439490ba3ec7ecde07f8de4
To: <sip:7100@webcall.collnetwork.net>;tag=z9hG4bKPj6ba04f9e81d5480696f0358c8a6ca70d
CSeq: 31340 INVITE
WWW-Authenticate: Digest realm="asterisk",nonce="1759580555/30c0c0882683b399733f4cdead6e3234",opaque="7a3942fb4af3824e",algorithm=MD5,qop="auth"
Server: Asterisk PBX 22.2.0~dfsg+~cs6.15.60671435-2
Content-Length: 0
<--- Received SIP request (402 bytes) from UDP:172.16.48.5:49517 --->
ACK sip:7100@webcall.collnetwork.net SIP/2.0
Via: SIP/2.0/UDP 172.16.48.5:49517;rport;branch=z9hG4bKPj6ba04f9e81d5480696f0358c8a6ca70d
Max-Forwards: 70
From: <sip:1605@webcall.collnetwork.net>;tag=d80dfdedf439490ba3ec7ecde07f8de4
To: <sip:7100@webcall.collnetwork.net>;tag=z9hG4bKPj6ba04f9e81d5480696f0358c8a6ca70d
Call-ID: 2795b75be8dc436a82b9c72a4a65dad0
CSeq: 31340 ACK
Content-Length: 0
<--- Received SIP request (1717 bytes) from UDP:172.16.48.5:49517 --->
INVITE sip:7100@webcall.collnetwork.net SIP/2.0
Via: SIP/2.0/UDP 172.16.48.5:49517;rport;branch=z9hG4bKPj9d51acaea1e34926b7a7b09d135506e2
Max-Forwards: 70
From: <sip:1605@webcall.collnetwork.net>;tag=d80dfdedf439490ba3ec7ecde07f8de4
To: <sip:7100@webcall.collnetwork.net>
Contact: <sip:1605@172.16.48.5:49517;ob>
Call-ID: 2795b75be8dc436a82b9c72a4a65dad0
CSeq: 31341 INVITE
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 1800
Min-SE: 90
User-Agent: MicroSIP/3.22.0
Authorization: Digest username="1605", realm="asterisk", nonce="1759580555/30c0c0882683b399733f4cdead6e3234", uri="sip:7100@webcall.collnetwork.net", response="e1868ff528f33181850241fe42a0f4a3", algorithm=MD5, cnonce="e1ad4b0babea413c88d82080ced7a591", opaque="7a3942fb4af3824e", qop=auth, nc=00000001
Content-Type: application/sdp
Content-Length: 766

v=0
o=- 3968572955 3968572955 IN IP4 172.16.48.5
s=pjmedia
b=AS:352
t=0 0
a=X-nat:0
m=audio 4000 RTP/AVP 8 0 101
c=IN IP4 172.16.48.5
b=TIAS:64000
a=rtcp:4001 IN IP4 172.16.48.5
a=sendrecv
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ssrc:351082781 cname:3ea53a1e0f3760d1
m=video 4002 RTP/AVP 99 98 100 101
c=IN IP4 172.16.48.5
b=TIAS:256000
a=rtcp:4003 IN IP4 172.16.48.5
a=sendrecv
a=rtpmap:99 H264/90000
a=fmtp:99 profile-level-id=42e01e; packetization-mode=1
a=rtpmap:98 H263-1998/90000
a=fmtp:98 CIF=1;QCIF=1
a=rtpmap:100 VP8/90000
a=fmtp:100 max-fr=30; max-fs=580
a=rtpmap:101 VP9/90000
a=fmtp:101 max-fr=30; max-fs=580
a=ssrc:45757132 cname:3ea53a1e0f3760d1
a=rtcp-fb:* nack pli
<--- Transmitting SIP response (395 bytes) to UDP:172.16.48.5:49517 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.16.48.5:49517;rport=49517;received=172.16.48.5;branch=z9hG4bKPj9d51acaea1e34926b7a7b09d135506e2
Call-ID: 2795b75be8dc436a82b9c72a4a65dad0
From: <sip:1605@webcall.collnetwork.net>;tag=d80dfdedf439490ba3ec7ecde07f8de4
To: <sip:7100@webcall.collnetwork.net>
CSeq: 31341 INVITE
Server: Asterisk PBX 22.2.0~dfsg+~cs6.15.60671435-2
Content-Length: 0
<--- Transmitting SIP request (1801 bytes) to UDP:172.17.67.14:5060 --->
INVITE sip:7100@cucm-pub-west.fuotuoke.edu.ng:5060 SIP/2.0
Via: SIP/2.0/UDP 172.16.48.3:5060;rport;branch=z9hG4bKPj30ddd672-b1d1-4b64-8acf-3e32ddf08cfb
From: <sip:1605@172.16.48.3>;tag=0105cb4e-f9a4-46ff-96a9-92f841867966
To: <sip:7100@cucm-pub-west.fuotuoke.edu.ng>
Contact: <sip:asterisk@172.16.48.3:5060>
Call-ID: af112f8f-78d2-474f-8c69-950c089584b7
CSeq: 23029 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, INFO, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub, histinfo
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: Asterisk PBX 22.2.0~dfsg+~cs6.15.60671435-2
Content-Type: application/sdp
Content-Length: 1080

v=0
o=- 1023029259 1023029259 IN IP4 172.16.48.3
s=Asterisk
c=IN IP4 172.16.48.3
t=0 0
a=group:BUNDLE audio-0 video-1
m=audio 14690 RTP/AVPF 0 101
a=ice-ufrag:5039e50e34b2fb513b0446250c6f6268
a=ice-pwd:2a4fadac47e9d3283b434465668d7bfa
a=candidate:Hac103003 1 UDP 2130706431 172.16.48.3 14690 typ host
a=candidate:S665a67d4 1 UDP 1694498815 102.90.103.212 40232 typ srflx raddr 172.16.48.3 rport 14690
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:140
a=sendrecv
a=rtcp-mux
a=ssrc:825511479 cname:4da1597b-fb6c-4a5d-89e9-dcf6854ef794
a=mid:audio-0
m=video 12022 RTP/AVPF 99
a=ice-ufrag:6d15b1f602ffee5e5217c04b56b8513b
a=ice-pwd:607322b93775f2670ce8c15858d27e47
a=candidate:Hac103003 1 UDP 2130706431 172.16.48.3 12022 typ host
a=candidate:S665a67d4 1 UDP 1694498815 102.90.103.212 40233 typ srflx raddr 172.16.48.3 rport 12022
a=rtpmap:99 H264/90000
a=fmtp:99 packetization-mode=1;profile-level-id=42E01E
a=sendrecv
a=rtcp-mux
a=ssrc:1815478733 cname:e53475a4-5f6b-490c-90bf-286dbce43269
a=mid:video-1
<--- Received SIP response (380 bytes) from UDP:172.17.67.14:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.16.48.3:5060;rport;branch=z9hG4bKPj30ddd672-b1d1-4b64-8acf-3e32ddf08cfb
From: <sip:1605@172.16.48.3>;tag=0105cb4e-f9a4-46ff-96a9-92f841867966
To: <sip:7100@cucm-pub-west.fuotuoke.edu.ng>
Date: Sat, 04 Oct 2025 12:22:35 GMT
Call-ID: af112f8f-78d2-474f-8c69-950c089584b7
CSeq: 23029 INVITE
Allow-Events: presence
Content-Length: 0
<--- Received SIP response (964 bytes) from UDP:172.17.67.14:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 172.16.48.3:5060;rport;branch=z9hG4bKPj30ddd672-b1d1-4b64-8acf-3e32ddf08cfb
From: <sip:1605@172.16.48.3>;tag=0105cb4e-f9a4-46ff-96a9-92f841867966
To: <sip:7100@cucm-pub-west.fuotuoke.edu.ng>;tag=25951~7eaefdf2-baae-484c-8b86-bd5e78e7d13d-24699038
Date: Sat, 04 Oct 2025 12:22:35 GMT
Call-ID: af112f8f-78d2-474f-8c69-950c089584b7
CSeq: 23029 INVITE
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
Allow-Events: presence
Server: Cisco-CUCM12.5
Call-Info: <sip:172.17.67.14:5060>;method="NOTIFY;Event=telephone-event;Duration=500"
Supported: X-cisco-srtp-fallback
Supported: Geolocation
Session-ID: a9d5cb672711e2910c3e7bba24699040;remote=afab1261b068a5da39a307fe6ab25951
P-Asserted-Identity: "8001" <sip:9007@fuotuoke.edu.ng>
Remote-Party-ID: "8001" <sip:9007@fuotuoke.edu.ng>;party=called;screen=yes;privacy=off
Contact: <sip:7100@172.17.67.14:5060>
Content-Length: 0
<--- Transmitting SIP response (587 bytes) to UDP:172.16.48.5:49517 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 172.16.48.5:49517;rport=49517;received=172.16.48.5;branch=z9hG4bKPj9d51acaea1e34926b7a7b09d135506e2
Call-ID: 2795b75be8dc436a82b9c72a4a65dad0
From: <sip:1605@webcall.collnetwork.net>;tag=d80dfdedf439490ba3ec7ecde07f8de4
To: <sip:7100@webcall.collnetwork.net>;tag=5af3b612-0704-4a94-a194-830098a925e4
CSeq: 31341 INVITE
Server: Asterisk PBX 22.2.0~dfsg+~cs6.15.60671435-2
Contact: <sip:172.16.48.3:5060>
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, INFO, MESSAGE, REFER
Content-Length: 0
<--- Received SIP response (1503 bytes) from UDP:172.17.67.14:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.48.3:5060;rport;branch=z9hG4bKPj30ddd672-b1d1-4b64-8acf-3e32ddf08cfb
From: <sip:1605@172.16.48.3>;tag=0105cb4e-f9a4-46ff-96a9-92f841867966
To: <sip:7100@cucm-pub-west.fuotuoke.edu.ng>;tag=25951~7eaefdf2-baae-484c-8b86-bd5e78e7d13d-24699038
Date: Sat, 04 Oct 2025 12:22:35 GMT
Call-ID: af112f8f-78d2-474f-8c69-950c089584b7
CSeq: 23029 INVITE
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
Allow-Events: presence, kpml
Supported: replaces
Server: Cisco-CUCM12.5
Call-Info: <sip:172.17.67.14:5060>;method="NOTIFY;Event=telephone-event;Duration=500"
Supported: X-cisco-srtp-fallback
Supported: Geolocation
Session-Expires: 1800;refresher=uas
Require: timer
Session-ID: a9d5cb672711e2910c3e7bba24699040;remote=afab1261b068a5da39a307fe6ab25951
P-Asserted-Identity: "FUOCARE" <sip:9007@fuotuoke.edu.ng>
Remote-Party-ID: "FUOCARE" <sip:9007@fuotuoke.edu.ng>;party=called;screen=yes;privacy=off
Contact: <sip:7100@172.17.67.14:5060>
Content-Type: application/sdp
Content-Length: 423

v=0
o=CiscoSystemsCCM-SIP 25951 1 IN IP4 172.17.67.14
s=SIP Call
c=IN IP4 172.17.67.14
b=TIAS:64000
b=CT:64
b=AS:80
t=0 0
m=audio 24706 RTP/AVP 0 101
a=ptime:20
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=mid:0
m=video 0 RTP/AVP 31 34 96 97
a=rtpmap:31 H261/90000
a=rtpmap:34 H263/90000
a=rtpmap:96 H263-1998/90000
a=rtpmap:97 H264/90000
a=content:main
a=inactive
a=mid:0
<--- Transmitting SIP request (468 bytes) to UDP:172.17.67.14:5060 --->
ACK sip:7100@172.17.67.14:5060 SIP/2.0
Via: SIP/2.0/UDP 172.16.48.3:5060;rport;branch=z9hG4bKPj707500c9-f6ca-457b-8d67-40333e267097
From: <sip:1605@172.16.48.3>;tag=0105cb4e-f9a4-46ff-96a9-92f841867966
To: <sip:7100@cucm-pub-west.fuotuoke.edu.ng>;tag=25951~7eaefdf2-baae-484c-8b86-bd5e78e7d13d-24699038
Call-ID: af112f8f-78d2-474f-8c69-950c089584b7
CSeq: 23029 ACK
Max-Forwards: 70
User-Agent: Asterisk PBX 22.2.0~dfsg+~cs6.15.60671435-2
Content-Length: 0
<--- Transmitting SIP response (1155 bytes) to UDP:172.16.48.5:49517 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.48.5:49517;rport=49517;received=172.16.48.5;branch=z9hG4bKPj9d51acaea1e34926b7a7b09d135506e2
Call-ID: 2795b75be8dc436a82b9c72a4a65dad0
From: <sip:1605@webcall.collnetwork.net>;tag=d80dfdedf439490ba3ec7ecde07f8de4
To: <sip:7100@webcall.collnetwork.net>;tag=5af3b612-0704-4a94-a194-830098a925e4
CSeq: 31341 INVITE
Server: Asterisk PBX 22.2.0~dfsg+~cs6.15.60671435-2
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, INFO, MESSAGE, REFER
Contact: <sip:172.16.48.3:5060>
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800;refresher=uac
Require: timer
Content-Type: application/sdp
Content-Length: 438

v=0
o=- 3968572955 3968572957 IN IP4 172.16.48.3
s=Asterisk
c=IN IP4 172.16.48.3
t=0 0
m=audio 14468 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=maxptime:140
a=sendrecv
m=video 12834 RTP/AVP 99 100
a=rtpmap:99 H264/90000
a=fmtp:99 packetization-mode=1;profile-level-id=42E01E
a=rtpmap:100 VP8/90000
a=fmtp:100 max-fr=30;max-fs=580
a=sendrecv
<--- Received SIP request (385 bytes) from UDP:172.16.48.5:49517 --->
ACK sip:172.16.48.3:5060 SIP/2.0
Via: SIP/2.0/UDP 172.16.48.5:49517;rport;branch=z9hG4bKPj38533f0f9cf9420b868380630c4faec3
Max-Forwards: 70
From: <sip:1605@webcall.collnetwork.net>;tag=d80dfdedf439490ba3ec7ecde07f8de4
To: <sip:7100@webcall.collnetwork.net>;tag=5af3b612-0704-4a94-a194-830098a925e4
Call-ID: 2795b75be8dc436a82b9c72a4a65dad0
CSeq: 31341 ACK
Content-Length: 0
<--- Received SIP request (1126 bytes) from UDP:172.16.48.5:49517 --->
UPDATE sip:172.16.48.3:5060 SIP/2.0
Via: SIP/2.0/UDP 172.16.48.5:49517;rport;branch=z9hG4bKPj1c398d5b0d204ddaae0d2d1116aa2099
Max-Forwards: 70
From: <sip:1605@webcall.collnetwork.net>;tag=d80dfdedf439490ba3ec7ecde07f8de4
To: <sip:7100@webcall.collnetwork.net>;tag=5af3b612-0704-4a94-a194-830098a925e4
Contact: <sip:1605@172.16.48.5:49517;ob>
Call-ID: 2795b75be8dc436a82b9c72a4a65dad0
CSeq: 31342 UPDATE
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 1800;refresher=uac
Min-SE: 90
Content-Type: application/sdp
Content-Length: 562

v=0
o=- 3968572955 3968572956 IN IP4 172.16.48.5
s=pjmedia
b=AS:352
t=0 0
a=X-nat:0
m=audio 4000 RTP/AVP 0 101
c=IN IP4 172.16.48.5
b=TIAS:64000
a=rtcp:4001 IN IP4 172.16.48.5
a=ssrc:351082781 cname:3ea53a1e0f3760d1
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv
m=video 4002 RTP/AVP 99
c=IN IP4 172.16.48.5
b=TIAS:256000
a=rtcp:4003 IN IP4 172.16.48.5
a=ssrc:45757132 cname:3ea53a1e0f3760d1
a=rtcp-fb:* nack pli
a=rtpmap:99 H264/90000
a=fmtp:99 profile-level-id=42e01e; packetization-mode=1
a=sendrecv
<--- Transmitting SIP response (1070 bytes) to UDP:172.16.48.5:49517 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.48.5:49517;rport=49517;received=172.16.48.5;branch=z9hG4bKPj1c398d5b0d204ddaae0d2d1116aa2099
Call-ID: 2795b75be8dc436a82b9c72a4a65dad0
From: <sip:1605@webcall.collnetwork.net>;tag=d80dfdedf439490ba3ec7ecde07f8de4
To: <sip:7100@webcall.collnetwork.net>;tag=5af3b612-0704-4a94-a194-830098a925e4
CSeq: 31342 UPDATE
Session-Expires: 1800;refresher=uac
Require: timer
Contact: <sip:172.16.48.3:5060>
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, INFO, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub
Server: Asterisk PBX 22.2.0~dfsg+~cs6.15.60671435-2
Content-Type: application/sdp
Content-Length: 353

v=0
o=- 3968572955 3968572958 IN IP4 172.16.48.3
s=Asterisk
c=IN IP4 172.16.48.3
t=0 0
m=audio 14468 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:140
a=sendrecv
m=video 12834 RTP/AVP 99
a=rtpmap:99 H264/90000
a=fmtp:99 packetization-mode=1;profile-level-id=42E01E
a=sendrecv
<--- Transmitting SIP request (1008 bytes) to UDP:172.16.48.5:49517 --->
INVITE sip:1605@172.16.48.5:49517;ob SIP/2.0
Via: SIP/2.0/UDP 172.16.48.3:5060;rport;branch=z9hG4bKPjd0820361-cccb-4b89-a04e-ce84b373be85
From: <sip:7100@webcall.collnetwork.net>;tag=5af3b612-0704-4a94-a194-830098a925e4
To: <sip:1605@webcall.collnetwork.net>;tag=d80dfdedf439490ba3ec7ecde07f8de4
Contact: <sip:172.16.48.3:5060>
Call-ID: 2795b75be8dc436a82b9c72a4a65dad0
CSeq: 24179 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, INFO, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub, histinfo
Session-Expires: 1800;refresher=uas
Min-SE: 90
Max-Forwards: 70
User-Agent: Asterisk PBX 22.2.0~dfsg+~cs6.15.60671435-2
Content-Type: application/sdp
Content-Length: 257

v=0
o=- 3968572955 3968572959 IN IP4 172.16.48.3
s=Asterisk
c=IN IP4 172.16.48.3
t=0 0
m=audio 14468 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:140
a=sendrecv
m=video 0 RTP/AVP 99
<--- Received SIP response (1007 bytes) from UDP:172.16.48.5:49517 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.48.3:5060;rport=5060;received=172.16.48.3;branch=z9hG4bKPjd0820361-cccb-4b89-a04e-ce84b373be85
Call-ID: 2795b75be8dc436a82b9c72a4a65dad0
From: <sip:7100@webcall.collnetwork.net>;tag=5af3b612-0704-4a94-a194-830098a925e4
To: <sip:1605@webcall.collnetwork.net>;tag=d80dfdedf439490ba3ec7ecde07f8de4
CSeq: 24179 INVITE
Session-Expires: 1800;refresher=uas
Contact: <sip:1605@172.16.48.5:49517;ob>
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Supported: replaces, 100rel, timer, norefersub, trickle-ice
Content-Type: application/sdp
Content-Length: 354

v=0
o=- 3968572955 3968572957 IN IP4 172.16.48.5
s=pjmedia
b=AS:84
t=0 0
a=X-nat:0
m=audio 4000 RTP/AVP 0 101
c=IN IP4 172.16.48.5
b=TIAS:64000
a=rtcp:4001 IN IP4 172.16.48.5
a=sendrecv
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ssrc:351082781 cname:3ea53a1e0f3760d1
m=video 0 RTP/AVP 99
c=IN IP4 127.0.0.1
<--- Transmitting SIP request (454 bytes) to UDP:172.16.48.5:49517 --->
ACK sip:1605@172.16.48.5:49517;ob SIP/2.0
Via: SIP/2.0/UDP 172.16.48.3:5060;rport;branch=z9hG4bKPj30410817-dfef-48d0-911d-86690ea0d47d
From: <sip:7100@webcall.collnetwork.net>;tag=5af3b612-0704-4a94-a194-830098a925e4
To: <sip:1605@webcall.collnetwork.net>;tag=d80dfdedf439490ba3ec7ecde07f8de4
Call-ID: 2795b75be8dc436a82b9c72a4a65dad0
CSeq: 24179 ACK
Max-Forwards: 70
User-Agent: Asterisk PBX 22.2.0~dfsg+~cs6.15.60671435-2
Content-Length: 0
<--- Transmitting SIP request (2008 bytes) to UDP:172.17.67.14:5060 --->
INVITE sip:7100@172.17.67.14:5060 SIP/2.0
Via: SIP/2.0/UDP 172.16.48.3:5060;rport;branch=z9hG4bKPj0f0a5dc0-affe-44c0-914f-a085317d3672
From: <sip:1605@172.16.48.3>;tag=0105cb4e-f9a4-46ff-96a9-92f841867966
To: <sip:7100@cucm-pub-west.fuotuoke.edu.ng>;tag=25951~7eaefdf2-baae-484c-8b86-bd5e78e7d13d-24699038
Contact: <sip:asterisk@172.16.48.3:5060>
Call-ID: af112f8f-78d2-474f-8c69-950c089584b7
CSeq: 23030 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, INFO, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub, histinfo
Session-Expires: 1800;refresher=uas
Min-SE: 90
Max-Forwards: 70
User-Agent: Asterisk PBX 22.2.0~dfsg+~cs6.15.60671435-2
Content-Type: application/sdp
Content-Length: 1234

v=0
o=- 1023029259 1023029260 IN IP4 172.16.48.3
s=Asterisk
c=IN IP4 172.16.48.3
t=0 0
a=group:BUNDLE audio-0 video-1
m=audio 14690 RTP/AVPF 0 101
a=ice-ufrag:5039e50e34b2fb513b0446250c6f6268
a=ice-pwd:2a4fadac47e9d3283b434465668d7bfa
a=candidate:Hac103003 1 UDP 2130706431 172.16.48.3 14690 typ host
a=candidate:S665a67d4 1 UDP 1694498815 102.90.103.212 40232 typ srflx raddr 172.16.48.3 rport 14690
a=candidate:Hac103003 2 UDP 2130706430 172.16.48.3 14691 typ host
a=candidate:S665a67d4 2 UDP 1694498814 102.90.103.21

I may have missed something in fixing the markup.

Topic		Replies	Views
Video is being activated systematically in audio call Asterisk SIP	13	1895	March 23, 2021
Auto provision endpoints in Pjsip conf Asterisk Endpoints	11	154	October 4, 2025
[POSSIBLE BUG] Can't perform video calls with video from only one side Asterisk SIP	15	1354	August 8, 2022
Not able to get Video between 2 Asterisk Server Trunk setup Asterisk SIP	3	434	March 3, 2023
Help for Asterisk configurations with video intercom Asterisk SIP	5	1024	June 8, 2022

No video from Asterisk endpoint to CUCM ivr via sip trunk

Related topics