Poor video quality through Asterisk 1.8 how to fix?

Question

linux

Hello. On the aster through h264 I get a video resolution of 117 by 115 pixels in 64 kbps. Where to fix, where to look?

debug

-- Executing [[email protected]:1] Dial("SIP/2830-00012bdd", "SIP/2658") in new stack
  == Using SIP VIDEO CoS mark 6
  == Using SIP RTP CoS mark 5
Audio is at 19262
Video is at 227.210.15.12:15676
Adding codec 0x8 (alaw) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding video codec 0x200000 (h264) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 192.168.8.94:20495:
INVITE sip:[email protected]:20495;transport=TCP;ob SIP/2.0
Via: SIP/2.0/TCP 227.210.15.12:5060;branch=z9hG4bK3cdd68fb
Max-Forwards: 70
From: "2830" <sip:[email protected]>;tag=as0009a159
To: <sip:[email protected]:20495;transport=TCP;ob>
Contact: <sip:[email protected]:5060;transport=TCP>
Call-ID: [email protected]:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Mon, 13 Jan 2020 13:08:24 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 355

v=0
o=root 563775145 563775145 IN IP4 227.210.15.12
s=Asterisk PBX 1.8.18.0
c=IN IP4 227.210.15.12
b=CT:2000
t=0 0
m=audio 19262 RTP/AVP 8 0 3 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
m=video 15676 RTP/AVP 99
a=rtpmap:99 H264/90000
a=sendrecv

---
    -- Called SIP/2658

<--- SIP read from TCP:192.168.8.94:20495 --->
SIP/2.0 100 Trying
Via: SIP/2.0/TCP 227.210.15.12:5060;received=227.210.15.12;branch=z9hG4bK3cdd68fb
Call-ID: [email protected]:5060
From: "2830" <sip:[email protected]>;tag=as0009a159
To: <sip:[email protected];ob>
CSeq: 102 INVITE
Content-Length:  0


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

<--- SIP read from TCP:192.168.8.94:20495 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/TCP 227.210.15.12:5060;received=227.210.15.12;branch=z9hG4bK3cdd68fb
Call-ID: [email protected]:5060
From: "2830" <sip:[email protected]>;tag=as0009a159
To: <sip:[email protected];ob>;tag=d1691ce29b344f11aabe640e0b3bb4f8
CSeq: 102 INVITE
Contact: <sip:[email protected]:20495;transport=TCP;ob>
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Content-Length:  0


<------------->
--- (9 headers 0 lines) ---
list_route: hop: <sip:[email protected]:20495;transport=TCP;ob>
    -- SIP/2658-00012bde is ringing
    -- SIP/gorod-00012bdc answered SIP/2867-00012bdb
    -- Locally bridging SIP/2867-00012bdb and SIP/gorod-00012bdc

<--- SIP read from TCP:192.168.8.94:20495 --->
SIP/2.0 200 OK
Via: SIP/2.0/TCP 227.210.15.12:5060;received=227.210.15.12;branch=z9hG4bK3cdd68fb
Call-ID: [email protected]:5060
From: "2830" <sip:[email protected]>;tag=as0009a159
To: <sip:[email protected];ob>;tag=d1691ce29b344f11aabe640e0b3bb4f8
CSeq: 102 INVITE
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Contact: <sip:[email protected]:20495;transport=TCP;ob>
Supported: replaces, 100rel, timer, norefersub
Content-Type: application/sdp
Content-Length:   552

v=0
o=- 3787920505 3787920506 IN IP4 192.168.8.94
s=pjmedia
b=AS:3759
t=0 0
a=X-nat:0
m=audio 17 RTP/AVP 8 101
c=IN IP4 192.168.8.94
b=TIAS:64000
a=rtcp:18 IN IP4 192.168.8.94
a=sendrecv
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ssrc:1341676614 cname:272516430de56f3c
m=video 19 RTP/AVP 99
c=IN IP4 192.168.8.94
b=TIAS:3500000
a=rtcp:20 IN IP4 192.168.8.94
a=sendrecv
a=rtpmap:99 H264/90000
a=fmtp:99 profile-level-id=42000a; packetization-mode=0
a=ssrc:1214918660 cname:272516430de56f3c

<------------->
--- (11 headers 23 lines) ---
Found RTP audio format 8
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 101
Found RTP video format 99
Found video description format H264 for ID 99
Capabilities: us - 0x20000e (gsm|ulaw|alaw|h264), peer - audio=0x8 (alaw)/video=0x200000 (h264)/text=0x0 (nothing), combined - 0x200008 (alaw|h264)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.8.94:17
Peer video RTP is at port 192.168.8.94:19
list_route: hop: <sip:[email protected]:20495;transport=TCP;ob>
set_destination: Parsing <sip:[email protected]:20495;transport=TCP;ob> for address/port to send to
set_destination: set destination to 192.168.8.94:20495
Transmitting (no NAT) to 192.168.8.94:20495:
ACK sip:[email protected]:20495;transport=TCP;ob SIP/2.0
Via: SIP/2.0/TCP 227.210.15.12:5060;branch=z9hG4bK437d5ded
Max-Forwards: 70
From: "2830" <sip:[email protected]>;tag=as0009a159
To: <sip:[email protected]:20495;transport=TCP;ob>;tag=d1691ce29b344f11aabe640e0b3bb4f8
Contact: <sip:[email protected]:5060;transport=TCP>
Call-ID: [email protected]:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0


---
    -- SIP/2658-00012bde answered SIP/2830-00012bdd
    -- Locally bridging SIP/2830-00012bdd and SIP/2658-00012bde

<--- SIP read from TCP:192.168.8.94:20495 --->
INFO sip:[email protected]:5060;transport=TCP SIP/2.0
Via: SIP/2.0/TCP 192.168.8.94:20495;rport;branch=z9hG4bKPj1ebc49481a684a2fba5fccfe1fdded8d;alias
Max-Forwards: 70
From: <sip:[email protected];ob>;tag=d1691ce29b344f11aabe640e0b3bb4f8
To: "2830" <sip:[email protected]>;tag=as0009a159
Call-ID: [email protected]:5060
CSeq: 24389 INFO
User-Agent: MicroSIP/3.19.19
Content-Type: application/media_control+xml
Content-Length:   146

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

<--- Transmitting (no NAT) to 192.168.8.94:20495 --->
SIP/2.0 200 OK
Via: SIP/2.0/TCP 192.168.8.94:20495;rport;branch=z9hG4bKPj1ebc49481a684a2fba5fccfe1fdded8d;alias;received=192.168.8.94
From: <sip:[email protected];ob>;tag=d1691ce29b344f11aabe640e0b3bb4f8
o: "2830" <sip:[email protected]>;tag=as0009a159
Call-ID: [email protected]:5060
CSeq: 24389 INFO
Server: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>
set_destination: Parsing <sip:[email protected]:20495;transport=TCP;ob> for address/port to send to
set_destination: set destination to 192.168.8.94:20495
Reliably Transmitting (no NAT) to 192.168.8.94:20495:
INFO sip:[email protected]:20495;transport=TCP;ob SIP/2.0
Via: SIP/2.0/TCP 227.210.15.12:5060;branch=z9hG4bK7a00797a
Max-Forwards: 70
From: "2830" <sip:[email protected]>;tag=as0009a159
To: <sip:[email protected]:20495;transport=TCP;ob>;tag=d1691ce29b344f11aabe640e0b3bb4f8
Contact: <sip:[email protected]:5060;transport=TCP>
Call-ID: [email protected]:5060
CSeq: 103 INFO
User-Agent: Asterisk PBX
Content-Type: application/media_control+xml
Content-Length: 205

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

---

<--- SIP read from TCP:192.168.8.94:20495 --->
SIP/2.0 200 OK
Via: SIP/2.0/TCP 227.210.15.12:5060;received=227.210.15.12;branch=z9hG4bK7a00797a
Call-ID: [email protected]:5060
From: "2830" <sip:[email protected]>;tag=as0009a159
To: <sip:[email protected];ob>;tag=d1691ce29b344f11aabe640e0b3bb4f8
CSeq: 103 INFO
Content-Length:  0


<------------->
--- (7 headers 0 lines) ---
    -- Executing [[email protected]:1] Congestion("SIP/2867-00012bdb", "1") in new stack
  == Spawn extension (trunkdial, h, 1) exited non-zero on 'SIP/2867-00012bdb'
  == Spawn extension (trunkdial, 92900786, 1) exited non-zero on 'SIP/2867-00012bdb'
    -- Executing [[email protected]:1] Congestion("SIP/2830-00012bdd", "1") in new stack
  == Spawn extension (office, h, 1) exited non-zero on 'SIP/2830-00012bdd'
Scheduling destruction of SIP dialog '[email protected]:5060' in 32000 ms (Method: INFO)
set_destination: Parsing <sip:[email protected]:20495;transport=TCP;ob> for address/port to send to
set_destination: set destination to 192.168.8.94:20495
Reliably Transmitting (no NAT) to 192.168.8.94:20495:
BYE sip:[email protected]:20495;transport=TCP;ob SIP/2.0
Via: SIP/2.0/TCP 227.210.15.12:5060;branch=z9hG4bK1bf2125a
Max-Forwards: 70
From: "2830" <sip:[email protected]>;tag=as0009a159
To: <sip:[email protected]:20495;transport=TCP;ob>;tag=d1691ce29b344f11aabe640e0b3bb4f8
Call-ID: [email protected]:5060
CSeq: 104 BYE
User-Agent: Asterisk PBX
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


---
  == Spawn extension (office, 2658, 1) exited non-zero on 'SIP/2830-00012bdd'

<--- SIP read from TCP:192.168.8.94:20495 --->
SIP/2.0 200 OK
Via: SIP/2.0/TCP 227.210.15.12:5060;received=227.210.15.12;branch=z9hG4bK1bf2125a
Call-ID: [email protected]:5060
From: "2830" <sip:[email protected]>;tag=as0009a159
To: <sip:[email protected];ob>;tag=d1691ce29b344f11aabe640e0b3bb4f8
CSeq: 104 BYE
Content-Length:  0


<------------->
--- (7 headers 0 lines) ---
Really destroying SIP dialog '[email protected]:5060' Method: INFO
pbx*CLI> sip set debug off
SIP Debugging Disabled
pbx*CLI> exit

Answer 1

Send a signal through PJSIP

Answer 2

Try to set
maxcallbitrate=10000 in sip.conf
Or some other higher value and check the result
. Asterisk itself cannot transcode video, so you need to debug the protocol and look for why end devices negotiate such permission through asterisk. By googling, I saw that the problem is popular, but I did not see the exact solution.