No video from Asterisk endpoint to CUCM ivr via sip trunk

collinks79 · October 4, 2025, 3:38pm

Hello everyone ,
I set up sip trunk between Asterisk and CUCM via sip Trunk.
CUCM endpoint can call Asterisk endpoint and vice versa . Both video and audio work .They were not working before . I played around with the codex and they started working
I have cti route point configured in CUCM . I configured this as _7xxx in the Asterisk dial plan. When Asterisk dials 7100, the CUCM triggers the uccx application which plays the ivr sound to the Asterisk endpoint . When the uccx Agent picks or uccx ivr redirects the call to another CUCM end point , video will be disabled on CUCM endpoint (either uccx Agent phone or another CUCM endpoint that received the transferred call)

On the other hand, when CUCM endpoint calls the same ivr at 7100, and uccx ivr transfers the call to a microsip registered on Asterisk server ,video will work
My little observation based on sip trace
When Asterisk calls CUCM ivr at 7100 it seems video is not established. As a result when that call is transferred or an Agent picks that call, video will be disabled
Conversely ,if the CUCM endpoint calls the same CUCM ivr ,video is established
What could disable the video when Asterisk endpoint calls CUCM ivr
See sip trace below

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](http://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](http://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](http://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](http://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](http://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](http://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](http://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

COLLINS ONYEGBADO | B.Tech; Msc | CCNA; CCNP
Head, Hardware Maintenance Operations & Networking Unit.
Mobile: +2348064550911 | Voip:4001 | www.fuotuoke.edu.ng
ICT CENTER
FEDERAL UNIVERSITY OTUOKE, BYELSA STATE

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.

collinks79 · October 4, 2025, 5:03pm

Based on some troubleshooting so far
When Asterisk endpoint calls CUCM endpoint , there will be video . When CUCM endpoint transfers to another CUCM endpoint ,video will be disabled

Also, when Asterisk endpoint calls CUCM endpoint ,video world well. When that CUCM endpoint places call on hold and resume ,there will no video

When Asterisk calls CUCM endpoint and places that call on hold and resumes ,there will be video

I suspect CUCM SDP renegotiation with Asterisk

COLLINS ONYEGBADO | B.Tech; Msc | CCNA; CCNP
Head, Hardware Maintenance Operations & Networking Unit.
Mobile: +2348064550911 | Voip:4001 | www.fuotuoke.edu.ng
ICT CENTER
FEDERAL UNIVERSITY OTUOKE, BYELSA STATE

On Sat, Oct 4, 2025, 17:58 david551 via Asterisk Community <notifications@asterisk.discoursemail.com> wrote:

	david551 October 4

PJSIP Logging enabled

<--- Received SIP request (374 bytes) from UDP:[172.17.67.14:5060](http://172.17.67.14:5060) --->
OPTIONS sip:[172.16.48.3:5060](http://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](mailto:c9755f80-1f11bf1f-3e89-e4311ac@172.17.67.14)
User-Agent: Cisco-CUCM12.5
CSeq: 101 OPTIONS
Contact: <sip:[172.17.67.14:5060](http://172.17.67.14:5060)>
Max-Forwards: 0
Content-Length: 0
<--- Transmitting SIP response (838 bytes) to UDP:[172.17.67.14:5060](http://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](mailto: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](http://172.16.48.5:49517) --->
INVITE [sip:7100@webcall.collnetwork.net](mailto:sip%3A7100@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](mailto:sip%3A1605@webcall.collnetwork.net)>;tag=d80dfdedf439490ba3ec7ecde07f8de4
To: <[sip:7100@webcall.collnetwork.net](mailto:sip%3A7100@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](http://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](mailto:sip%3A1605@webcall.collnetwork.net)>;tag=d80dfdedf439490ba3ec7ecde07f8de4
To: <[sip:7100@webcall.collnetwork.net](mailto:sip%3A7100@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](http://172.16.48.5:49517) --->
ACK [sip:7100@webcall.collnetwork.net](mailto:sip%3A7100@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](mailto:sip%3A1605@webcall.collnetwork.net)>;tag=d80dfdedf439490ba3ec7ecde07f8de4
To: <[sip:7100@webcall.collnetwork.net](mailto:sip%3A7100@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](http://172.16.48.5:49517) --->
INVITE [sip:7100@webcall.collnetwork.net](mailto:sip%3A7100@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](mailto:sip%3A1605@webcall.collnetwork.net)>;tag=d80dfdedf439490ba3ec7ecde07f8de4
To: <[sip:7100@webcall.collnetwork.net](mailto:sip%3A7100@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](mailto:sip%3A7100@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](http://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](mailto:sip%3A1605@webcall.collnetwork.net)>;tag=d80dfdedf439490ba3ec7ecde07f8de4
To: <[sip:7100@webcall.collnetwork.net](mailto:sip%3A7100@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](http://172.17.67.14:5060) --->
INVITE [sip:7100@cucm-pub-west.fuotuoke.edu.ng:5060](http://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](mailto:sip%3A1605@172.16.48.3)>;tag=0105cb4e-f9a4-46ff-96a9-92f841867966
To: <[sip:7100@cucm-pub-west.fuotuoke.edu.ng](mailto:sip%3A7100@cucm-pub-west.fuotuoke.edu.ng)>
Contact: <[sip:asterisk@172.16.48.3:5060](http://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](http://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](mailto:sip%3A1605@172.16.48.3)>;tag=0105cb4e-f9a4-46ff-96a9-92f841867966
To: <[sip:7100@cucm-pub-west.fuotuoke.edu.ng](mailto:sip%3A7100@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](http://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](mailto:sip%3A1605@172.16.48.3)>;tag=0105cb4e-f9a4-46ff-96a9-92f841867966
To: <[sip:7100@cucm-pub-west.fuotuoke.edu.ng](mailto:sip%3A7100@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](http://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](mailto:sip%3A9007@fuotuoke.edu.ng)>
Remote-Party-ID: "8001" <[sip:9007@fuotuoke.edu.ng](mailto:sip%3A9007@fuotuoke.edu.ng)>;party=called;screen=yes;privacy=off
Contact: <[sip:7100@172.17.67.14:5060](http://sip:7100@172.17.67.14:5060)>
Content-Length: 0
<--- Transmitting SIP response (587 bytes) to UDP:[172.16.48.5:49517](http://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](mailto:sip%3A1605@webcall.collnetwork.net)>;tag=d80dfdedf439490ba3ec7ecde07f8de4
To: <[sip:7100@webcall.collnetwork.net](mailto:sip%3A7100@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](http://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](http://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](mailto:sip%3A1605@172.16.48.3)>;tag=0105cb4e-f9a4-46ff-96a9-92f841867966
To: <[sip:7100@cucm-pub-west.fuotuoke.edu.ng](mailto:sip%3A7100@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](http://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](mailto:sip%3A9007@fuotuoke.edu.ng)>
Remote-Party-ID: "FUOCARE" <[sip:9007@fuotuoke.edu.ng](mailto:sip%3A9007@fuotuoke.edu.ng)>;party=called;screen=yes;privacy=off
Contact: <[sip:7100@172.17.67.14:5060](http://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](http://172.17.67.14:5060) --->
ACK [sip:7100@172.17.67.14:5060](http://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](mailto:sip%3A1605@172.16.48.3)>;tag=0105cb4e-f9a4-46ff-96a9-92f841867966
To: <[sip:7100@cucm-pub-west.fuotuoke.edu.ng](mailto:sip%3A7100@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](http://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](mailto:sip%3A1605@webcall.collnetwork.net)>;tag=d80dfdedf439490ba3ec7ecde07f8de4
To: <[sip:7100@webcall.collnetwork.net](mailto:sip%3A7100@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](http://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](http://172.16.48.5:49517) --->
ACK sip:[172.16.48.3:5060](http://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](mailto:sip%3A1605@webcall.collnetwork.net)>;tag=d80dfdedf439490ba3ec7ecde07f8de4
To: <[sip:7100@webcall.collnetwork.net](mailto:sip%3A7100@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](http://172.16.48.5:49517) --->
UPDATE sip:[172.16.48.3:5060](http://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](mailto:sip%3A1605@webcall.collnetwork.net)>;tag=d80dfdedf439490ba3ec7ecde07f8de4
To: <[sip:7100@webcall.collnetwork.net](mailto:sip%3A7100@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](http://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](mailto:sip%3A1605@webcall.collnetwork.net)>;tag=d80dfdedf439490ba3ec7ecde07f8de4
To: <[sip:7100@webcall.collnetwork.net](mailto:sip%3A7100@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](http://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](http://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](mailto:sip%3A7100@webcall.collnetwork.net)>;tag=5af3b612-0704-4a94-a194-830098a925e4
To: <[sip:1605@webcall.collnetwork.net](mailto:sip%3A1605@webcall.collnetwork.net)>;tag=d80dfdedf439490ba3ec7ecde07f8de4
Contact: <sip:[172.16.48.3:5060](http://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](http://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](mailto:sip%3A7100@webcall.collnetwork.net)>;tag=5af3b612-0704-4a94-a194-830098a925e4
To: <[sip:1605@webcall.collnetwork.net](mailto:sip%3A1605@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](http://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](mailto:sip%3A7100@webcall.collnetwork.net)>;tag=5af3b612-0704-4a94-a194-830098a925e4
To: <[sip:1605@webcall.collnetwork.net](mailto:sip%3A1605@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](http://172.17.67.14:5060) --->
INVITE [sip:7100@172.17.67.14:5060](http://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](mailto:sip%3A1605@172.16.48.3)>;tag=0105cb4e-f9a4-46ff-96a9-92f841867966
To: <[sip:7100@cucm-pub-west.fuotuoke.edu.ng](mailto:sip%3A7100@cucm-pub-west.fuotuoke.edu.ng)>;tag=25951~7eaefdf2-baae-484c-8b86-bd5e78e7d13d-24699038
Contact: <[sip:asterisk@172.16.48.3:5060](http://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.

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david551 · October 4, 2025, 5:20pm

The Cisco has refused the video offer, towards 7100, from Asterisk, by setting the port number to zero and also setting the state to inactive. It has subsequently done an UPDATE to enable the video stream. Asterisk seems to accept this, but then does a re-INVITE, towards 1605, to delete the stream. It then does a re-INVITE towards the 7100, but the log appears to end, abruptly, halfway through this, so I don’t know what was done with the video stream.

There is no evidence of a transfer, and there are no timestamps, to give clues. The latter is usually the result of screen scraping the console log, rather than using the actual log file.

As far as I know, if you refuse a stream, that stream no longer exists, so I think the UPDATE is creating a new stream and Asterisk doesn’t handle new video streams being created mid-call.

david551 · October 4, 2025, 5:30pm

Correction

The update is from 1605, so 7100 refuses the video stream and that is an end for it.

collinks79 · October 4, 2025, 5:46pm

thanks a lot for your contribution. The sip trace below shows 1603 from Asterisk calling cisco jabber on 0001. On the 0001 endpoint on cucm,i placed call on hold and resumed but no video.only Audio.webcall*CLI> pjsip set logger on
PJSIP Logging enabled
<— Received SIP request (1491 bytes) from UDP:172.16.48.245:12966 —>
INVITE sip:0001@172.16.48.3 SIP/2.0
Via: SIP/2.0/UDP 172.16.48.245:12966;branch=z9hG4bK-524287-1—e61dda52e7481a7d;rport
Max-Forwards: 70
Contact: <sip:1604@172.16.48.245:12966>;+sip.instance=“urn:uuid:c57ee66e-5a18-39ef-86aa-b7a9e6380e98”
Call-ID: hXM7PyZfbPiWqdbEYduljw..
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, REGISTER, SUBSCRIBE, INFO, PUBLISH, UPDATE
Content-Type: application/sdp
Supported: replaces, answermode, eventlist, park-info, outbound, path
User-Agent: PortSIP UC Client Android - v13.0.1
Allow-Events: hold, talk, conference, dialog, park-info
Content-Length: 774

v=0
o=- 3894553234156473122 2 IN IP4 172.16.48.245
s=-
t=0 0
m=audio 11500 RTP/AVP 8 0 101
c=IN IP4 172.16.48.245
a=mid:audio
a=sendrecv
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=ssrc:3340528121 cname:ZxPQOcLP7e0V3vTa
m=video 11502 RTP/AVP 125 120
c=IN IP4 172.16.48.245
a=mid:video
a=sendrecv
a=rtcp-rsize
a=rtpmap:125 H264/90000
a=rtcp-fb:125 goog-remb
a=rtcp-fb:125 transport-cc
a=rtcp-fb:125 ccm fir
a=rtcp-fb:125 nack
a=rtcp-fb:125 nack pli
a=fmtp:125 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=42001f
a=rtpmap:120 VP8/90000
a=rtcp-fb:120 goog-remb
a=rtcp-fb:120 transport-cc
a=rtcp-fb:120 ccm fir
a=rtcp-fb:120 nack
a=rtcp-fb:120 nack pli
a=ssrc:2348145795 cname:ZxPQOcLP7e0V3vTa

<— Transmitting SIP response (532 bytes) to UDP:172.16.48.245:12966 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.16.48.245:12966;rport=12966;received=172.16.48.245;branch=z9hG4bK-524287-1—e61dda52e7481a7d
Call-ID: hXM7PyZfbPiWqdbEYduljw..
CSeq: 1 INVITE
WWW-Authenticate: Digest realm=“asterisk”,nonce=“1759599685/d3411a91fb1b17cd230a2d926fb471b0”,opaque=“52e6296e176c2667”,algorithm=MD5,qop=“auth”
Server: Asterisk PBX 22.2.0~dfsg+~cs6.15.60671435-2
Content-Length: 0

<— Received SIP request (325 bytes) from UDP:172.16.48.245:12966 —>
ACK sip:0001@172.16.48.3 SIP/2.0
Via: SIP/2.0/UDP 172.16.48.245:12966;branch=z9hG4bK-524287-1—e61dda52e7481a7d;rport
Max-Forwards: 70
Call-ID: hXM7PyZfbPiWqdbEYduljw..
CSeq: 1 ACK
Content-Length: 0

<— Received SIP request (1773 bytes) from UDP:172.16.48.245:12966 —>
INVITE sip:0001@172.16.48.3 SIP/2.0
Via: SIP/2.0/UDP 172.16.48.245:12966;branch=z9hG4bK-524287-1—2246ed035826722f;rport
Max-Forwards: 70
Contact: <sip:1604@172.16.48.245:12966>;+sip.instance=“urn:uuid:c57ee66e-5a18-39ef-86aa-b7a9e6380e98”
Call-ID: hXM7PyZfbPiWqdbEYduljw..
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, REGISTER, SUBSCRIBE, INFO, PUBLISH, UPDATE
Content-Type: application/sdp
Supported: replaces, answermode, eventlist, park-info, outbound, path
User-Agent: PortSIP UC Client Android - v13.0.1
Authorization: Digest username=“1604”,realm=“asterisk”,nonce=“1759599685/d3411a91fb1b17cd230a2d926fb471b0”,uri=“sip:0001@172.16.48.3”,response=“d9897492442f3966bd519c7b49ddcd25”,cnonce=“2e95434d931799cd8e57cd46fa293f64”,nc=00000001,qop=auth,algorithm=MD5,opaque=“52e6296e176c2667”
Allow-Events: hold, talk, conference, dialog, park-info
Content-Length: 774

v=0
o=- 3894553234156473122 2 IN IP4 172.16.48.245
s=-
t=0 0
m=audio 11500 RTP/AVP 8 0 101
c=IN IP4 172.16.48.245
a=mid:audio
a=sendrecv
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=ssrc:3340528121 cname:ZxPQOcLP7e0V3vTa
m=video 11502 RTP/AVP 125 120
c=IN IP4 172.16.48.245
a=mid:video
a=sendrecv
a=rtcp-rsize
a=rtpmap:125 H264/90000
a=rtcp-fb:125 goog-remb
a=rtcp-fb:125 transport-cc
a=rtcp-fb:125 ccm fir
a=rtcp-fb:125 nack
a=rtcp-fb:125 nack pli
a=fmtp:125 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=42001f
a=rtpmap:120 VP8/90000
a=rtcp-fb:120 goog-remb
a=rtcp-fb:120 transport-cc
a=rtcp-fb:120 ccm fir
a=rtcp-fb:120 nack
a=rtcp-fb:120 nack pli
a=ssrc:2348145795 cname:ZxPQOcLP7e0V3vTa

<— Transmitting SIP response (340 bytes) to UDP:172.16.48.245:12966 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.16.48.245:12966;rport=12966;received=172.16.48.245;branch=z9hG4bK-524287-1—2246ed035826722f
Call-ID: hXM7PyZfbPiWqdbEYduljw..
CSeq: 2 INVITE
Server: Asterisk PBX 22.2.0~dfsg+~cs6.15.60671435-2
Content-Length: 0

<— Transmitting SIP request (1340 bytes) to UDP:172.17.67.14:5060 —>
INVITE sip:0001@cucm-pub-west.fuotuoke.edu.ng:5060 SIP/2.0
Via: SIP/2.0/UDP 172.16.48.3:5060;rport;branch=z9hG4bKPj0983223e-6315-4d56-b40d-10f5005045b3
Contact: <sip:asterisk@172.16.48.3:5060>
Call-ID: 6de10687-7047-4cbf-be52-fb86a8fb5e96
CSeq: 25567 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: 612

v=0
o=- 916175609 916175609 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 11712 RTP/AVPF 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
a=rtcp-mux
a=ssrc:1372488929 cname:287aa228-5fc7-41e4-ac99-4326fc207ecc
a=mid:audio-0
m=video 16540 RTP/AVPF 99
a=rtpmap:99 H264/90000
a=fmtp:99 packetization-mode=1;level-asymmetry-allowed=1;profile-level-id=42001F
a=sendrecv
a=rtcp-mux
a=ssrc:273600798 cname:c2c9afc8-0ce2-4d04-a47c-15f74367d510
a=mid:video-1

<— Received SIP response (387 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=z9hG4bKPj0983223e-6315-4d56-b40d-10f5005045b3

david551 · October 4, 2025, 5:54pm

Please mark up logs as preformatted text!

Asterisk offers video. The B party provisionally acknowledges receipt of the request, but nothing further is logged; no Ringing; no call acceptance. It looks like the B side has broken during the call set up.

The provisional response doesn’t contain any SDP, so there is no indication as to whether the video stream would be accepted,

This log shows a problem definitely outside of Asterisk. The previous log could also show that, or could show that the video offer was considered unacceptable by the downstream system.

collinks79 · October 5, 2025, 10:41am

Ok. I agree with you that the Major issue is from the CUCM. When Asterisk endpoint calls CUCM ivr, CUCM has video disabled which makes Asterisk to mark it’s video port 0. When the Agent behind the IVR picks the call , CUCM endpoint of the Agent does not see video and stays disabled . However ,if I manually start the video on the CUCM endpoint , Asterisk will respond with the video.
This might be applicable to the hold/resume/transfer issue where video is disabled when it involves Asterisk endpoint
What baffles me is that All the CUCM endpoints are not experiencing this issue. It’s only Asterisk endpoint and CUCM endpoints. Among Asterisk endpoints, this issue does not happen
So it lies on what connects Asterisk to CUCM and what connects CUCM to Asterisk

COLLINS ONYEGBADO | B.Tech; Msc | CCNA; CCNP
Head, Hardware Maintenance Operations & Networking Unit.
Mobile: +2348064550911 | Voip:4001 | www.fuotuoke.edu.ng
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On Sat, Oct 4, 2025, 19:06 david551 via Asterisk Community <notifications@asterisk.discoursemail.com> wrote:

david551
October 4

Please mark up logs as preformatted text!

Asterisk offers video. The B party provisionally acknowledges receipt of the request, but nothing further is logged; no Ringing; no call acceptance. It looks like the B side has broken during the call set up.

The provisional response doesn’t contain any SDP, so there is no indication as to whether the video stream would be accepted,

This log shows a problem definitely outside of Asterisk. The previous log could also show that, or could show that the video offer as considered unacceptable by the downstream system.

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david551 · October 5, 2025, 11:01am

It is CUCM that is sending SDP with port 0, and that doesn’t just disable video, it deletes the video stream.

If the stream is coming back it is really as a new stream. I thought that Asterisk couldn’t add a new video stream, but it is possible that feature has been added recently.

TedM · October 6, 2025, 2:30am

But that is the normal, expected case. Few people running a UCM at a business use the IVR for internal extension-to-extension calls. Almost all use the IVR for handling incoming calls from the PSTN that will land on internal extensions. The PSTN does not handle video calls only audio and if you send an audio call to a videophone and enable video, then the videophone presents a black screen on it’s video screen which then makes all of the soft buttons the phone COULD be displaying on that screen, useless.

If you make an audio only call from an extension on your Asterisk system to an extension with a video phone, what happens? Does the phone try displaying a video screen? When I do it from a non-video CP-6941 to a video CP-8941 the videophone does NOT attempt to display video.

If you transfer a call through the IVR from a videophone on the CUCM to another videophone on the CUCM does it display video?

collinks79 · October 6, 2025, 5:28am

Thank you for your contribution. I understand that most people use ivr for internal extension. If a Cisco video phone calls the same IVR which does not send video, the Cisco video phone will have its phone disabled but when the Agent who uses video phone picks, the Cisco video phone will enable it’s video . Does this make sense ?
I expected the same same behavior from Asterisk video endpoint which calls the same ivr and disables it’s video like what the Cisco video phone does when it calls ivr. so when the Agent picks, there should have been SDP renegotiation because the Agent’s video phone is sending video stream. Since Asterisk has preserved it’s video port 0, the Agent’s video phone stays disabled
I want to look at the sip trace from CUCM and see what happens.
The same issue happens when Asterisk video phone calls CUCM video phone over a sip trunk. Video will work on both directions. When Cisco video phone places the call on hold and resumes there will be no video. This tells us that possibly Cisco video phone is not sending video in the re-invite or that Asterisk is not performing video renegotiation with the CUCM . Conversely ,if Asterisk endpoint is the one that places the call on hold and resumes the call, video will work

PSTN is definitely an audio which reaches the IVR via external PSTN line . One wouldn’t expect video from an audio enabled phone.
In my case ,this is Asterisk that supports both video and Audio over a sip trunk to another sip server (CUCM) who also support both video and audio . Assuming I install another CUCM and connects it over a sip trunk to this main CUCM , do you think if an endpoint from the new CUCM calls the ivr in the main CUCM , video will be disabled if the Agent picks the call?

COLLINS ONYEGBADO | B.Tech; Msc | CCNA; CCNP
Head, Hardware Maintenance Operations & Networking Unit.
Mobile: +2348064550911 | Voip:4001 | www.fuotuoke.edu.ng
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On Mon, Oct 6, 2025, 03:41 TedM via Asterisk Community <notifications@asterisk.discoursemail.com> wrote:

TedM
October 6

collinks79:

When Asterisk endpoint calls CUCM ivr, CUCM has video disabled which makes Asterisk to mark it’s video port 0.

But that is the normal, expected case. Few people running a UCM at a business use the IVR for internal extension-to-extension calls. Almost all use the IVR for handling incoming calls from the PSTN that will land on internal extensions. The PSTN does not handle video calls only audio and if you send an audio call to a videophone and enable video, then the videophone presents a black screen on it’s video screen which then makes all of the soft buttons the phone COULD be displaying on that screen, useless.

If you make an audio only call from an extension on your Asterisk system to an extension with a video phone, what happens? Does the phone try displaying a video screen? When I do it from a non-video CP-6941 to a video CP-8941 the videophone does NOT attempt to display video.

If you transfer a call through the IVR from a videophone on the CUCM to another videophone on the CUCM does it display video?

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david551 · October 6, 2025, 7:49am

In the logs you have provided so far, CUCM is completely deleting the video stream, on the B side, not just placing it on hold.

TedM · October 7, 2025, 2:58pm

I don’t know because when plugging 2 CUCM’s together you get a choice of building “regular” SIP trunks or “enhanced” ones that (presumably) are Cisco-proprietary and give you extra things. I’d assume the enhanced trunks would work and the regulars would not.

The issue I think is whether or not the call is entirely processed through the PBX. When you have a case with an endpoint registered into Asterisk and an endpoint registered into the CUCM and PBX-to-PBX trunks tying them together, my observation is when the call is connected it stays routed through the PBXes and PBX-to-PBX trunk. When it’s 2 endpoints registered into Asterisk and a call is connected, the call goes direct from phone to phone, and the CUCM works the same way. But when it’s a call into the IVR then transferred to an extension - who knows what happens. Your going to have to do some testing. If the IVR resides on Asterisk and it’s 2 extensions registered into Asterisk and they call each other via the IVR - then does the call end up phone-to-phone or phone-to-Asterisk-to-phone? It seems you have many test cases here you will need to chase down and see what happens.

collinks79 · October 7, 2025, 7:43pm

Thank you for your contribution. Definitely it will work . Like you mentioned,it might be Cisco Proprietary related .
I am exploring possible options resolve the issues
CUCM CTI route point is not a physical endpoint and might not send video in in the initial sip invite. To isolate this issue, I decided to call directly from CUCM endpoint to Asterisk endpoint. I placed the call on hold and resumed the call . To my greatest surprise ,the video didn’t display . However , when I toggled on the video button on the Cisco endpoint ,the video displayed . This tells me that CUCM endpoint sent a new stream which trigger Asterisk to negotiate with the stream

There is this experiment I did a year ago when I was looking for an open source sip that would support webrtc. I came across janus sip gateway. There is an online demo where you can enter sip url and dial it.
I was able to dial my extension 0001@fuotuoke.edu.ng. The call routed via my expressway edge and core to my CUCM . I can’t remember exactly if I dialed the CUCM ivr during the experiment
I could have tried that experiment again but I don’t have a public IP address

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Head, Hardware Maintenance Operations & Networking Unit.
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On Tue, Oct 7, 2025, 16:09 TedM via Asterisk Community <notifications@asterisk.discoursemail.com> wrote:

TedM
October 7

collinks79:

Assuming I install another CUCM and connects it over a sip trunk to this main CUCM , do you think if an endpoint from the new CUCM calls the ivr in the main CUCM , video will be disabled if the Agent picks the call?

I don’t know because when plugging 2 CUCM’s together you get a choice of building “regular” SIP trunks or “enhanced” ones that (presumably) are Cisco-proprietary and give you extra things. I’d assume the enhanced trunks would work and the regulars would not.

The issue I think is whether or not the call is entirely processed through the PBX. When you have a case with an endpoint registered into Asterisk and an endpoint registered into the CUCM and PBX-to-PBX trunks tying them together, my observation is when the call is connected it stays routed through the PBXes and PBX-to-PBX trunk. When it’s 2 endpoints registered into Asterisk and a call is connected, the call goes direct from phone to phone, and the CUCM works the same way. But when it’s a call into the IVR then transferred to an extension - who knows what happens. Your going to have to do some testing. If the IVR resides on Asterisk and it’s 2 extensions registered into Asterisk and they call each other via the IVR - then does the call end up phone-to-phone or phone-to-Asterisk-to-phone? It seems you have many test cases here you will need to chase down and see what happens.

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system · November 6, 2025, 7:43pm

This topic was automatically closed 30 days after the last reply. New replies are no longer allowed.

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No video from Asterisk endpoint to CUCM ivr via sip trunk

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