Asterisk 22.4.X issue with video

Asterisk Asterisk SIP
1

Hello,

I’m having a problem with the Grandstream GDS3710 video intercom and a GSC3570 trying to view video. The scenario is:

GSC3570 (Ext. 3001) → Asterisk01 → Asterisk02 → Asterisk03 → GDS3710 (Ext. 451)

In the GSC3570’s INVITE, I see that the SDP sends:

m=video 5006 RTP/AVP 99 100 101 34 100
b=AS:2048
a=sendrecv
a=rtpmap:99 H264/90000
a=fmtp:99 profile-level-id=42801F; packetization-mode=1
a=rtpmap:100 H264/90000
a=fmtp:100 profile-level-id=4D001F; packetization-mode=1
a=rtpmap:101 H264/90000
a=fmtp:101 profile-level-id=64001F; packetization-mode=1
a=rtpmap:34 H263/90000
a=fmtp:34 CIF=2; QCIF=2
a=rtpmap:100 H263-1998/90000
a=fmtp:100 CIF=2; QCIF=2

When Asterisk01 communicates with Asterisk02, it removes the video profile (profile-level-id) and only sends to Asterisk02:

m=video 10812 RTP/AVP 99
a=rtpmap:99 H264/90000
a=sendrecv

In all elements (endpoints, SIP trunk) allow=h264, direct_media=no is enabled:

[Asterisk02-aor]
type = aor
max_contacts = 2
remove_existing = yes
contact = sip:10.0.3.3:5073
qualify_frequency = 120
qualify_timeout = 30

[PBX_OPS.01-to-Asterisk02]
type = endpoint
context = pbx01
allow = !all,g722,alaw,ulaw,h264
aors = Asterisk02-aor
direct_media = no
transport = transport-udp-PBXhomelab
accept_multiple_sdp_answers = yes

[PBXa_OPS.01-identify]
type = identify
match = 10.0.3.3:5070
endpoint = PBX_OPS.01-to-Asterisk02


[3001]
accept_multiple_sdp_answers = yes
auth = 3001-auth
aors = 3001
callerid = Caller <3001>
direct_media = no
mailboxes = 3001@buzones
allow_subscribe = no
transport = transport-udp
type = endpoint
context = css7
allow = !all,g722,alaw,ulaw,g729,vp8,h264,vp9
trust_id_outbound = yes
device_state_busy_at = 1
dtmf_mode = rfc4733
language = es
tos_audio = ef
tos_video = af41

A solution on the Grandstream side (but it is still a patch for a problem that should not exist) is to check the “SIP Proxy Compatibility Mode” option.

Pjsip debug:

<--- Received SIP request (1448 bytes) from UDP:IP_GSC3570:5066 --->
INVITE sip:451@IP_Asterisk01 SIP/2.0
Via: SIP/2.0/UDP IP_GSC3570:5066;branch=z9hG4bK616170467;rport
From: "GSC3570.OPS.01" <sip:3001@IP_Asterisk01>;tag=1827722761
To: <sip:451@IP_Asterisk01>
Call-ID: 2096891238-5066-2@BJC.BGI.E.E
CSeq: 10 INVITE
Contact: "GSC3570.OPS.01" <sip:3001@IP_GSC3570:5066>
Max-Forwards: 70
User-Agent: Grandstream GSC3570 1.0.7.8
Privacy: none
P-Preferred-Identity: "GSC3570.OPS.01" <sip:3001@IP_Asterisk01>
P-Access-Network-Info: IEEE-EUI-48;eui-48-addr=MACaddress
P-Emergency-Info: IEEE-EUI-48;eui-48-addr=MACaddress
Supported: replaces, path, timer
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Type: application/sdp
Accept: application/sdp, application/dtmf-relay
Content-Length:   648

v=0
o=3001 8000 8000 IN IP4 IP_GSC3570
s=SIP Call
c=IN IP4 IP_GSC3570
t=0 0
m=audio 5004 RTP/AVP 9 8 101
a=sendrecv
a=rtpmap:9 G722/8000
a=ptime:20
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
m=video 5006 RTP/AVP 99 100 101 34 100
b=AS:2048
a=sendrecv
a=rtpmap:99 H264/90000
a=fmtp:99 profile-level-id=42801F; packetization-mode=1
a=rtpmap:100 H264/90000
a=fmtp:100 profile-level-id=4D001F; packetization-mode=1
a=rtpmap:101 H264/90000
a=fmtp:101 profile-level-id=64001F; packetization-mode=1
a=rtpmap:34 H263/90000
a=fmtp:34 CIF=2; QCIF=2
a=rtpmap:100 H263-1998/90000
a=fmtp:100 CIF=2; QCIF=2

<--- Transmitting SIP response (479 bytes) to UDP:IP_GSC3570:5066 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP IP_GSC3570:5066;rport=5066;received=IP_GSC3570;branch=z9hG4bK616170467
Call-ID: 2096891238-5066-2@BJC.BGI.E.E
From: "GSC3570.OPS.01" <sip:3001@IP_Asterisk01>;tag=1827722761
To: <sip:451@IP_Asterisk01>;tag=z9hG4bK616170467
CSeq: 10 INVITE
WWW-Authenticate: <NONCE>
Server: OPS PBX homelab
Content-Length:  0


<--- Received SIP request (286 bytes) from UDP:IP_GSC3570:5066 --->
ACK sip:451@IP_Asterisk01 SIP/2.0
Via: SIP/2.0/UDP IP_GSC3570:5066;branch=z9hG4bK616170467;rport
From: "GSC3570.OPS.01" <sip:3001@IP_Asterisk01>;tag=1827722761
To: <sip:451@IP_Asterisk01>;tag=z9hG4bK616170467
Call-ID: 2096891238-5066-2@BJC.BGI.E.E
CSeq: 10 ACK
Content-Length: 0


<--- Received SIP request (1715 bytes) from UDP:IP_GSC3570:5066 --->
INVITE sip:451@IP_Asterisk01 SIP/2.0
Via: SIP/2.0/UDP IP_GSC3570:5066;branch=z9hG4bK142189798;rport
From: "GSC3570.OPS.01" <sip:3001@IP_Asterisk01>;tag=1827722761
To: <sip:451@IP_Asterisk01>
Call-ID: 2096891238-5066-2@BJC.BGI.E.E
CSeq: 11 INVITE
Contact: "GSC3570.OPS.01" <sip:3001@IP_GSC3570:5066>
Authorization: <NONCE>
Max-Forwards: 70
User-Agent: Grandstream GSC3570 1.0.7.8
Privacy: none
P-Preferred-Identity: "GSC3570.OPS.01" <sip:3001@IP_Asterisk01>
P-Access-Network-Info: IEEE-EUI-48;eui-48-addr=MACaddress
P-Emergency-Info: IEEE-EUI-48;eui-48-addr=MACaddress
Supported: replaces, path, timer
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Type: application/sdp
Accept: application/sdp, application/dtmf-relay
Content-Length:   648

v=0
o=3001 8000 8000 IN IP4 IP_GSC3570
s=SIP Call
c=IN IP4 IP_GSC3570
t=0 0
m=audio 5004 RTP/AVP 9 8 101
a=sendrecv
a=rtpmap:9 G722/8000
a=ptime:20
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
m=video 5006 RTP/AVP 99 100 101 34 100
b=AS:2048
a=sendrecv
a=rtpmap:99 H264/90000
a=fmtp:99 profile-level-id=42801F; packetization-mode=1
a=rtpmap:100 H264/90000
a=fmtp:100 profile-level-id=4D001F; packetization-mode=1
a=rtpmap:101 H264/90000
a=fmtp:101 profile-level-id=64001F; packetization-mode=1
a=rtpmap:34 H263/90000
a=fmtp:34 CIF=2; QCIF=2
a=rtpmap:100 H263-1998/90000
a=fmtp:100 CIF=2; QCIF=2

<--- Transmitting SIP response (306 bytes) to UDP:IP_GSC3570:5066 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP IP_GSC3570:5066;rport=5066;received=IP_GSC3570;branch=z9hG4bK142189798
Call-ID: 2096891238-5066-2@BJC.BGI.E.E
From: "GSC3570.OPS.01" <sip:3001@IP_Asterisk01>;tag=1827722761
To: <sip:451@IP_Asterisk01>
CSeq: 11 INVITE
Server: OPS PBX homelab
Content-Length:  0
	
<--- Transmitting SIP request (1012 bytes) to UDP:IP_Asterisk02:5073 --->
INVITE sip:451@IP_Asterisk02:5073 SIP/2.0
Via: SIP/2.0/UDP IP_Asterisk01:5073;rport;branch=z9hG4bKPj0df9477a-f5b4-4866-856a-3dbda02397e9
From: "OPS homelab" <sip:3001@IP_Asterisk01>;tag=8e764100-585b-467e-a152-88002db55d9a
To: <sip:451@IP_Asterisk02>
Contact: <sip:asterisk@IP_Asterisk01:5073>
Call-ID: 4fcff944-490f-4c49-a350-39ca5c547efc
CSeq: 1820 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: OPS PBX homelab
Content-Type: application/sdp
Content-Length:   347

v=0
o=- 1325014821 1325014821 IN IP4 IP_Asterisk01
s=Asterisk
c=IN IP4 IP_Asterisk01
t=0 0
m=audio 10810 RTP/AVP 9 8 0 101
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
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 10814 RTP/AVP 99
a=rtpmap:99 H264/90000
a=sendrecv

<--- Received SIP response (357 bytes) from UDP:IP_Asterisk02:5073 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP IP_Asterisk01:5073;rport=5073;received=IP_Asterisk01;branch=z9hG4bKPj0df9477a-f5b4-4866-856a-3dbda02397e9
Call-ID: 4fcff944-490f-4c49-a350-39ca5c547efc
From: "OPS homelab" <sip:3001@IP_Asterisk01>;tag=8e764100-585b-467e-a152-88002db55d9a
To: <sip:451@IP_Asterisk02>
CSeq: 1820 INVITE
Server: OPS SBC
Content-Length:  0


<--- Received SIP response (546 bytes) from UDP:IP_Asterisk02:5073 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP IP_Asterisk01:5073;rport=5073;received=IP_Asterisk01;branch=z9hG4bKPj0df9477a-f5b4-4866-856a-3dbda02397e9
Call-ID: 4fcff944-490f-4c49-a350-39ca5c547efc
From: "OPS homelab" <sip:3001@IP_Asterisk01>;tag=8e764100-585b-467e-a152-88002db55d9a
To: <sip:451@IP_Asterisk02>;tag=3bee5136-ee19-4ae3-ad56-8377835268a5
CSeq: 1820 INVITE
Server: OPS SBC
Contact: <sip:IP_Asterisk02:5073>
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, INFO, MESSAGE, REFER
Content-Length:  0


    -- PJSIP/PBX_OPS.01-to-PBXhomelab01-00000060 is ringing
<--- Transmitting SIP response (499 bytes) to UDP:IP_GSC3570:5066 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP IP_GSC3570:5066;rport=5066;received=IP_GSC3570;branch=z9hG4bK142189798
Call-ID: 2096891238-5066-2@BJC.BGI.E.E
From: "GSC3570.OPS.01" <sip:3001@IP_Asterisk01>;tag=1827722761
To: <sip:451@IP_Asterisk01>;tag=09865586-53bb-4c36-8ac3-019600ef6fc9
CSeq: 11 INVITE
Server: OPS PBX homelab
Contact: <sip:IP_Asterisk01:5060>
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, INFO, MESSAGE, REFER
Content-Length:  0


<--- Received SIP response (1015 bytes) from UDP:IP_Asterisk02:5073 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP IP_Asterisk01:5073;rport=5073;received=IP_Asterisk01;branch=z9hG4bKPj0df9477a-f5b4-4866-856a-3dbda02397e9
Call-ID: 4fcff944-490f-4c49-a350-39ca5c547efc
From: "OPS homelab" <sip:3001@IP_Asterisk01>;tag=8e764100-585b-467e-a152-88002db55d9a
To: <sip:451@IP_Asterisk02>;tag=3bee5136-ee19-4ae3-ad56-8377835268a5
CSeq: 1820 INVITE
Server: OPS SBC
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, INFO, MESSAGE, REFER
Contact: <sip:IP_Asterisk02:5073>
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800;refresher=uac
Require: timer
Content-Type: application/sdp
Content-Length:   339

v=0
o=- 1325014821 1325014823 IN IP4 IP_Asterisk02
s=Asterisk
c=IN IP4 IP_Asterisk02
t=0 0
m=audio 20214 RTP/AVP 9 8 0 101
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
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 20194 RTP/AVP 99
a=rtpmap:99 H264/90000
a=sendrecv

       > 0x55c2786a72c0 -- Strict RTP learning after remote address set to: IP_Asterisk02:20214
       > 0x55c2783d9920 -- Strict RTP learning after remote address set to: IP_Asterisk02:20194
	   
<--- Transmitting SIP request (409 bytes) to UDP:IP_Asterisk02:5073 --->
ACK sip:IP_Asterisk02:5073 SIP/2.0
Via: SIP/2.0/UDP IP_Asterisk01:5073;rport;branch=z9hG4bKPjb2609d72-f870-42f4-8992-83183f163595
From: "OPS homelab" <sip:3001@IP_Asterisk01>;tag=8e764100-585b-467e-a152-88002db55d9a
To: <sip:451@IP_Asterisk02>;tag=3bee5136-ee19-4ae3-ad56-8377835268a5
Call-ID: 4fcff944-490f-4c49-a350-39ca5c547efc
CSeq: 1820 ACK
Max-Forwards: 70
User-Agent: OPS PBX homelab
Content-Length:  0


    -- PJSIP/PBX_OPS.01-to-PBXhomelab01-00000060 answered PJSIP/3001-0000005f
       > 0x55c2786805b0 -- Strict RTP learning after remote address set to: IP_GSC3570:5004
       > 0x55c2783d0d70 -- Strict RTP learning after remote address set to: IP_GSC3570:5006
<--- Transmitting SIP response (887 bytes) to UDP:IP_GSC3570:5066 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP IP_GSC3570:5066;rport=5066;received=IP_GSC3570;branch=z9hG4bK142189798
Call-ID: 2096891238-5066-2@BJC.BGI.E.E
From: "GSC3570.OPS.01" <sip:3001@IP_Asterisk01>;tag=1827722761
To: <sip:451@IP_Asterisk01>;tag=09865586-53bb-4c36-8ac3-019600ef6fc9
CSeq: 11 INVITE
Server: OPS PBX homelab
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, INFO, MESSAGE, REFER
Contact: <sip:IP_Asterisk01:5060>
Supported: 100rel, timer, replaces, norefersub
Content-Type: application/sdp
Content-Length:   311

v=0
o=- 8000 8002 IN IP4 IP_Asterisk01
s=Asterisk
c=IN IP4 IP_Asterisk01
t=0 0
m=audio 10812 RTP/AVP 9 8 101
a=rtpmap:9 G722/8000
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 10818 RTP/AVP 99
a=rtpmap:99 H264/90000
a=sendrecv

    -- Channel PJSIP/PBX_OPS.01-to-PBXhomelab01-00000060 joined 'simple_bridge' basic-bridge <b73c5059-b23d-4eb8-b1ca-d31677346371>
    -- Channel PJSIP/3001-0000005f joined 'simple_bridge' basic-bridge <b73c5059-b23d-4eb8-b1ca-d31677346371>
<--- Received SIP request (1043 bytes) from UDP:IP_Asterisk02:5070 --->
INVITE sip:asterisk@IP_Asterisk01:5073 SIP/2.0
Via: SIP/2.0/UDP IP_Asterisk02:5070;rport;branch=z9hG4bKPj62a18dfe-8047-451a-8cc6-0ebbbe2b8793
From: <sip:451@IP_Asterisk02>;tag=3bee5136-ee19-4ae3-ad56-8377835268a5
To: "OPS homelab" <sip:3001@IP_Asterisk01>;tag=8e764100-585b-467e-a152-88002db55d9a
Contact: <sip:IP_Asterisk02:5070>
Call-ID: 4fcff944-490f-4c49-a350-39ca5c547efc
CSeq: 4512 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: OPS SBC
Content-Type: application/sdp
Content-Length:   339

v=0
o=- 1325014821 1325014824 IN IP4 IP_Asterisk02
s=Asterisk
c=IN IP4 IP_Asterisk02
t=0 0
m=audio 20214 RTP/AVP 9 8 0 101
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
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 20194 RTP/AVP 99
a=rtpmap:99 H264/90000
a=sendonly

       > 0x55c2786a72c0 -- Strict RTP learning after remote address set to: IP_Asterisk02:20214
       > 0x55c2783d9920 -- Strict RTP learning after remote address set to: IP_Asterisk02:20194
<--- Transmitting SIP response (1020 bytes) to UDP:IP_Asterisk02:5070 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP IP_Asterisk02:5070;rport=5070;received=IP_Asterisk02;branch=z9hG4bKPj62a18dfe-8047-451a-8cc6-0ebbbe2b8793
Call-ID: 4fcff944-490f-4c49-a350-39ca5c547efc
From: <sip:451@IP_Asterisk02>;tag=3bee5136-ee19-4ae3-ad56-8377835268a5
To: "OPS homelab" <sip:3001@IP_Asterisk01>;tag=8e764100-585b-467e-a152-88002db55d9a
CSeq: 4512 INVITE
Session-Expires: 1800;refresher=uas
Contact: <sip:asterisk@IP_Asterisk01:5073>
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, INFO, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub
Server: OPS PBX homelab
Content-Type: application/sdp
Content-Length:   347

v=0
o=- 1325014821 1325014822 IN IP4 IP_Asterisk01
s=Asterisk
c=IN IP4 IP_Asterisk01
t=0 0
m=audio 10810 RTP/AVP 9 8 0 101
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
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 10814 RTP/AVP 99
a=rtpmap:99 H264/90000
a=recvonly

<--- Received SIP request (410 bytes) from UDP:IP_Asterisk02:5070 --->
ACK sip:asterisk@IP_Asterisk01:5073 SIP/2.0
Via: SIP/2.0/UDP IP_Asterisk02:5070;rport;branch=z9hG4bKPj03a286dd-1f92-424f-ab7d-ded65914eb80
From: <sip:451@IP_Asterisk02>;tag=3bee5136-ee19-4ae3-ad56-8377835268a5
To: "OPS homelab" <sip:3001@IP_Asterisk01>;tag=8e764100-585b-467e-a152-88002db55d9a
Call-ID: 4fcff944-490f-4c49-a350-39ca5c547efc
CSeq: 4512 ACK
Max-Forwards: 70
User-Agent: OPS SBC
Content-Length:  0


<--- Received SIP request (530 bytes) from UDP:IP_GSC3570:5066 --->
ACK sip:IP_Asterisk01:5060 SIP/2.0
Via: SIP/2.0/UDP IP_GSC3570:5066;branch=z9hG4bK1420986101;rport
From: "GSC3570.OPS.01" <sip:3001@IP_Asterisk01>;tag=1827722761
To: <sip:451@IP_Asterisk01>;tag=09865586-53bb-4c36-8ac3-019600ef6fc9
Call-ID: 2096891238-5066-2@BJC.BGI.E.E
CSeq: 11 ACK
Contact: <sip:3001@IP_GSC3570:5066>
Max-Forwards: 70
Supported: replaces, path, timer
User-Agent: Grandstream GSC3570 1.0.7.8
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0


<--- Transmitting SIP request (987 bytes) to UDP:IP_GSC3570:5066 --->
INVITE sip:3001@IP_GSC3570:5066 SIP/2.0
Via: SIP/2.0/UDP IP_Asterisk01:5060;rport;branch=z9hG4bKPj1cd473ff-b505-40ed-bf4e-a0c1c2764822
From: <sip:451@IP_Asterisk01>;tag=09865586-53bb-4c36-8ac3-019600ef6fc9
To: "GSC3570.OPS.01" <sip:3001@IP_Asterisk01>;tag=1827722761
Contact: <sip:IP_Asterisk01:5060>
Call-ID: 2096891238-5066-2@BJC.BGI.E.E
CSeq: 31569 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: OPS PBX homelab
Content-Type: application/sdp
Content-Length:   311

v=0
o=- 8000 8003 IN IP4 IP_Asterisk01
s=Asterisk
c=IN IP4 IP_Asterisk01
t=0 0
m=audio 10812 RTP/AVP 9 8 101
a=rtpmap:9 G722/8000
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 10818 RTP/AVP 99
a=rtpmap:99 H264/90000
a=sendonly

       > 0x55c2783d9920 -- Strict RTP switching to RTP target address IP_Asterisk02:20194 as source
       > 0x55c2786805b0 -- Strict RTP switching to RTP target address IP_GSC3570:5004 as source
<--- Received SIP response (537 bytes) from UDP:IP_GSC3570:5066 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP IP_Asterisk01:5060;rport=5060;branch=z9hG4bKPj1cd473ff-b505-40ed-bf4e-a0c1c2764822
From: <sip:451@IP_Asterisk01>;tag=09865586-53bb-4c36-8ac3-019600ef6fc9
To: "GSC3570.OPS.01" <sip:3001@IP_Asterisk01>;tag=1827722761
Call-ID: 2096891238-5066-2@BJC.BGI.E.E
CSeq: 31569 INVITE
Contact: <sip:3001@IP_GSC3570:5066>
Supported: replaces, path, timer
User-Agent: Grandstream GSC3570 1.0.7.8
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0


<--- Received SIP response (969 bytes) from UDP:IP_GSC3570:5066 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP IP_Asterisk01:5060;rport=5060;branch=z9hG4bKPj1cd473ff-b505-40ed-bf4e-a0c1c2764822
From: <sip:451@IP_Asterisk01>;tag=09865586-53bb-4c36-8ac3-019600ef6fc9
To: "GSC3570.OPS.01" <sip:3001@IP_Asterisk01>;tag=1827722761
Call-ID: 2096891238-5066-2@BJC.BGI.E.E
CSeq: 31569 INVITE
Contact: <sip:3001@IP_GSC3570:5066>
Supported: replaces, path, timer
User-Agent: Grandstream GSC3570 1.0.7.8
Session-Expires: 1800;refresher=uac
Require: timer
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Type: application/sdp
Content-Length:   348

v=0
o=3001 8000 8001 IN IP4 IP_GSC3570
s=SIP Call
c=IN IP4 IP_GSC3570
t=0 0
m=audio 5004 RTP/AVP 9 8 101
a=sendrecv
a=rtpmap:9 G722/8000
a=ptime:20
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
m=video 5006 RTP/AVP 99
a=recvonly
a=rtpmap:99 H264/90000
a=fmtp:99 sprop-parameter-sets=Z0KADJWgUH5A,aM4Ecg==

       > 0x55c2783d0d70 -- Strict RTP learning after remote address set to: IP_GSC3570:5006
<--- Transmitting SIP request (392 bytes) to UDP:IP_GSC3570:5066 --->
ACK sip:3001@IP_GSC3570:5066 SIP/2.0
Via: SIP/2.0/UDP IP_Asterisk01:5060;rport;branch=z9hG4bKPj4af13dff-3330-462a-8c87-3f4e24c4ab03
From: <sip:451@IP_Asterisk01>;tag=09865586-53bb-4c36-8ac3-019600ef6fc9
To: "GSC3570.OPS.01" <sip:3001@IP_Asterisk01>;tag=1827722761
Call-ID: 2096891238-5066-2@BJC.BGI.E.E
CSeq: 31569 ACK
Max-Forwards: 70
User-Agent: OPS PBX homelab
Content-Length:  0

       > 0x55c2786a72c0 -- Strict RTP switching to RTP target address IP_Asterisk02:20214 as source

<--- Received SIP request (529 bytes) from UDP:IP_GSC3570:5066 --->
BYE sip:IP_Asterisk01:5060 SIP/2.0
Via: SIP/2.0/UDP IP_GSC3570:5066;branch=z9hG4bK390611563;rport
From: "GSC3570.OPS.01" <sip:3001@IP_Asterisk01>;tag=1827722761
To: <sip:451@IP_Asterisk01>;tag=09865586-53bb-4c36-8ac3-019600ef6fc9
Call-ID: 2096891238-5066-2@BJC.BGI.E.E
CSeq: 12 BYE
Contact: <sip:3001@IP_GSC3570:5066>
Max-Forwards: 70
Supported: replaces, path, timer
User-Agent: Grandstream GSC3570 1.0.7.8
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0

<--- Transmitting SIP response (340 bytes) to UDP:IP_GSC3570:5066 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP IP_GSC3570:5066;rport=5066;received=IP_GSC3570;branch=z9hG4bK390611563
Call-ID: 2096891238-5066-2@BJC.BGI.E.E
From: "GSC3570.OPS.01" <sip:3001@IP_Asterisk01>;tag=1827722761
To: <sip:451@IP_Asterisk01>;tag=09865586-53bb-4c36-8ac3-019600ef6fc9
CSeq: 12 BYE
Server: OPS PBX homelab
Content-Length:  0

<--- Transmitting SIP request (433 bytes) to UDP:IP_Asterisk02:5070 --->
BYE sip:IP_Asterisk02:5070 SIP/2.0
Via: SIP/2.0/UDP IP_Asterisk01:5073;rport;branch=z9hG4bKPj0d948e0d-a2c1-49c8-a0bf-4e39dc9a28cf
From: "OPS homelab" <sip:3001@IP_Asterisk01>;tag=8e764100-585b-467e-a152-88002db55d9a
To: <sip:451@IP_Asterisk02>;tag=3bee5136-ee19-4ae3-ad56-8377835268a5
Call-ID: 4fcff944-490f-4c49-a350-39ca5c547efc
CSeq: 1821 BYE
Reason: Q.850;cause=16
Max-Forwards: 70
User-Agent: OPS PBX homelab
Content-Length:  0

<--- Received SIP response (391 bytes) from UDP:IP_Asterisk02:5070 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP IP_Asterisk01:5073;rport=5073;received=IP_Asterisk01;branch=z9hG4bKPj0d948e0d-a2c1-49c8-a0bf-4e39dc9a28cf
Call-ID: 4fcff944-490f-4c49-a350-39ca5c547efc
From: "OPS homelab" <sip:3001@IP_Asterisk01>;tag=8e764100-585b-467e-a152-88002db55d9a
To: <sip:451@IP_Asterisk02>;tag=3bee5136-ee19-4ae3-ad56-8377835268a5
CSeq: 1821 BYE
Server: OPS SBC
Content-Length:  0


sv01*CLI> pjsip set logger off
PJSIP Logging disabled

Thanks.
Regards,

2

The problem you are running into is Asterisk is a B2BUA (Back to Back User Agent) and not a true SIP Proxy.

There is no copying of SDP between legs, each call leg is it’s own based on the endpoint configuration.

3

The problem is definitely not with Asterisk; Grandstream seems to have its “proper implementation” with the intercoms and control stations. I’m discussing and testing with them through their support.

Thanks.

4

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