Help for Asterisk configurations with video intercom

Asterisk Asterisk SIP
1

Hi,

on my PC with Linux I have an application programmed in Python with PJSUA2 used to connect to my video intercom IP.

First I used an Android Application as Sip Server installed on my mobile phone, It words but it is not so comfortable.

Now I’m trying to use Asterisk installed on my PC because much more usable.

I’m not an expert and I followed my tutorial in order to understand the correct way to configure Asterisk. I succeed in making to ring the “virtual phone”on my PC and in opening and closing the call, but I couldn’t “transport” incoming and outcoming audio.

My firs idea was a conflict between the ports, so I installed Asterisk on another PC but the prblem wasn’t solved.

In PJSUA2 report, it seems that audio is closed because not used (?!)

pjsip.conf, extensions.conf, voicemail.conf and Asterisk and PJSUA2 report attached

someone can help me.
thanks

extensions.conf

[internal]
exten => VideoCitofono,1,Dial(PJSIP/VideoCitofono,60)
exten => VideoCitofono,n,Hangup()

exten => 1,1,Dial(PJSIP/1,60)
exten => 1,n,Hangup()

pjsip.conf

[transport]
type=transport
protocol=udp
bind=0.0.0.0
 
[VideoCitofono]
type = endpoint
context = internal
disallow = all
allow = ulaw
aors = VideoCitofono
auth = authVideoCitofono
 
[VideoCitofono]
type = aor
max_contacts=3
 
[authVideoCitofono]
type=auth
auth_type=userpass
password=VideoCitofono
username=VideoCitofono
 
[1]
type = endpoint
context = internal
disallow = all
allow = ulaw
aors = 1
auth = auth1
 
[1]
type = aor
max_contacts=3

[auth1]
type=auth
auth_type=userpass
password=1
username=1

voicemail.conf

[main]

VideoCitofono => VideoCitofono
1 => 1

pc_asterisk_debug(Call-_Respone).txt

Connected to Asterisk GIT-18-db824d8f6d currently running on marcello-S551LB (pid = 1360)
marcello-S551LB*CLI> pjsip set logger host 192.168.1.51
PJSIP Logging Enabled for host: 192.168.1.51
    -- Executing [1@internal:1] Dial("PJSIP/VideoCitofono-00000006", "PJSIP/1,60") in new stack
    -- Called PJSIP/1
<--- Transmitting SIP request (932 bytes) to UDP:192.168.1.51:46917 --->
INVITE sip:1@192.168.1.51:46917;ob SIP/2.0
Via: SIP/2.0/UDP 192.168.1.59:5060;rport;branch=z9hG4bKPj070dc9c9-1be3-402d-af9b-7ebafa8b95a9
From: "VideoCitofono" <sip:VideoCitofono@192.168.1.59>;tag=934f19ed-9993-456a-ad90-0a4d2f66e71a
To: <sip:1@192.168.1.51;ob>
Contact: <sip:asterisk@192.168.1.59:5060>
Call-ID: 387ff28f-c1c7-401a-bdd9-b6e8055ef4c8
CSeq: 31213 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub, histinfo
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: Asterisk PBX GIT-18-db824d8f6d
Content-Type: application/sdp
Content-Length:   235

v=0
o=- 594678824 594678824 IN IP4 192.168.1.59
s=Asterisk
c=IN IP4 192.168.1.59
t=0 0
m=audio 17738 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:150
a=sendrecv

<--- Received SIP response (357 bytes) from UDP:192.168.1.51:46917 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.59:5060;rport=5060;received=192.168.1.59;branch=z9hG4bKPj070dc9c9-1be3-402d-af9b-7ebafa8b95a9
Call-ID: 387ff28f-c1c7-401a-bdd9-b6e8055ef4c8
From: "VideoCitofono" <sip:VideoCitofono@192.168.1.59>;tag=934f19ed-9993-456a-ad90-0a4d2f66e71a
To: <sip:1@192.168.1.51;ob>
CSeq: 31213 INVITE
Content-Length:  0


<--- Received SIP response (537 bytes) from UDP:192.168.1.51:46917 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.1.59:5060;rport=5060;received=192.168.1.59;branch=z9hG4bKPj070dc9c9-1be3-402d-af9b-7ebafa8b95a9
Call-ID: 387ff28f-c1c7-401a-bdd9-b6e8055ef4c8
From: "VideoCitofono" <sip:VideoCitofono@192.168.1.59>;tag=934f19ed-9993-456a-ad90-0a4d2f66e71a
To: <sip:1@192.168.1.51;ob>;tag=87036ac9-5686-4602-b003-431c20432ac2
CSeq: 31213 INVITE
Contact: <sip:1@192.168.1.51:46917;ob>
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Content-Length:  0


    -- PJSIP/1-00000007 is ringing
<--- Received SIP response (983 bytes) from UDP:192.168.1.51:46917 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.59:5060;rport=5060;received=192.168.1.59;branch=z9hG4bKPj070dc9c9-1be3-402d-af9b-7ebafa8b95a9
Call-ID: 387ff28f-c1c7-401a-bdd9-b6e8055ef4c8
From: "VideoCitofono" <sip:VideoCitofono@192.168.1.59>;tag=934f19ed-9993-456a-ad90-0a4d2f66e71a
To: <sip:1@192.168.1.51;ob>;tag=87036ac9-5686-4602-b003-431c20432ac2
CSeq: 31213 INVITE
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Contact: <sip:1@192.168.1.51:46917;ob>
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 1800;refresher=uac
Require: timer
Content-Type: application/sdp
Content-Length:   316

v=0
o=- 3862623641 3862623642 IN IP4 192.168.1.51
s=pjmedia
b=AS:117
t=0 0
a=X-nat:0
m=audio 4006 RTP/AVP 0 101
c=IN IP4 192.168.1.51
b=TIAS:96000
a=rtcp:4007 IN IP4 192.168.1.51
a=sendrecv
a=rtpmap:0 PCMU/8000
a=ssrc:761689561 cname:3acf37564e7f7c23
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16

<--- Transmitting SIP request (451 bytes) to UDP:192.168.1.51:46917 --->
ACK sip:1@192.168.1.51:46917;ob SIP/2.0
Via: SIP/2.0/UDP 192.168.1.59:5060;rport;branch=z9hG4bKPj7c41309a-cb19-4728-aea9-625c7e13a7ae
From: "VideoCitofono" <sip:VideoCitofono@192.168.1.59>;tag=934f19ed-9993-456a-ad90-0a4d2f66e71a
To: <sip:1@192.168.1.51;ob>;tag=87036ac9-5686-4602-b003-431c20432ac2
Call-ID: 387ff28f-c1c7-401a-bdd9-b6e8055ef4c8
CSeq: 31213 ACK
Max-Forwards: 70
User-Agent: Asterisk PBX GIT-18-db824d8f6d
Content-Length:  0


    -- PJSIP/1-00000007 answered PJSIP/VideoCitofono-00000006
    -- Channel PJSIP/1-00000007 joined 'simple_bridge' basic-bridge <359b7385-704b-4def-a844-1300f6417282>
    -- Channel PJSIP/VideoCitofono-00000006 joined 'simple_bridge' basic-bridge <359b7385-704b-4def-a844-1300f6417282>
<--- Transmitting SIP request (986 bytes) to UDP:192.168.1.51:46917 --->
INVITE sip:1@192.168.1.51:46917;ob SIP/2.0
Via: SIP/2.0/UDP 192.168.1.59:5060;rport;branch=z9hG4bKPjd4a10d87-e8b8-43df-aeda-43762a01f5a3
From: "VideoCitofono" <sip:VideoCitofono@192.168.1.59>;tag=934f19ed-9993-456a-ad90-0a4d2f66e71a
To: <sip:1@192.168.1.51;ob>;tag=87036ac9-5686-4602-b003-431c20432ac2
Contact: <sip:asterisk@192.168.1.59:5060>
Call-ID: 387ff28f-c1c7-401a-bdd9-b6e8055ef4c8
CSeq: 31214 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub, histinfo
Session-Expires: 1800;refresher=uac
Min-SE: 90
Max-Forwards: 70
User-Agent: Asterisk PBX GIT-18-db824d8f6d
Content-Type: application/sdp
Content-Length:   234

v=0
o=- 594678824 594678825 IN IP4 192.168.1.59
s=Asterisk
c=IN IP4 192.168.1.65
t=0 0
m=audio 9654 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:150
a=sendrecv

Got  RTP packet from    192.168.1.65:9654 (type 00, seq 003418, ts 1069035052, len 000320)
Sent RTP packet to      192.168.1.51:4006 (type 00, seq 022710, ts 1069035048, len 000320)
Got  RTP packet from    192.168.1.65:9654 (type 00, seq 003419, ts 1069035371, len 000320)
Sent RTP packet to      192.168.1.51:4006 (type 00, seq 022711, ts 1069035368, len 000320)
<--- Received SIP response (983 bytes) from UDP:192.168.1.51:46917 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.59:5060;rport=5060;received=192.168.1.59;branch=z9hG4bKPjd4a10d87-e8b8-43df-aeda-43762a01f5a3
Call-ID: 387ff28f-c1c7-401a-bdd9-b6e8055ef4c8
From: "VideoCitofono" <sip:VideoCitofono@192.168.1.59>;tag=934f19ed-9993-456a-ad90-0a4d2f66e71a
To: <sip:1@192.168.1.51;ob>;tag=87036ac9-5686-4602-b003-431c20432ac2
CSeq: 31214 INVITE
Session-Expires: 1800;refresher=uac
Require: timer
Contact: <sip:1@192.168.1.51:46917;ob>
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Supported: replaces, 100rel, timer, norefersub
Content-Type: application/sdp
Content-Length:   316

v=0
o=- 3862623641 3862623643 IN IP4 192.168.1.51
s=pjmedia
b=AS:117
t=0 0
a=X-nat:0
m=audio 4006 RTP/AVP 0 101
c=IN IP4 192.168.1.51
b=TIAS:96000
a=rtcp:4007 IN IP4 192.168.1.51
a=sendrecv
a=rtpmap:0 PCMU/8000
a=ssrc:761689561 cname:3acf37564e7f7c23
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16

<--- Transmitting SIP request (451 bytes) to UDP:192.168.1.51:46917 --->
ACK sip:1@192.168.1.51:46917;ob SIP/2.0
Via: SIP/2.0/UDP 192.168.1.59:5060;rport;branch=z9hG4bKPjb37d0052-bbcb-421e-82c0-589f3c2302a6
From: "VideoCitofono" <sip:VideoCitofono@192.168.1.59>;tag=934f19ed-9993-456a-ad90-0a4d2f66e71a
To: <sip:1@192.168.1.51;ob>;tag=87036ac9-5686-4602-b003-431c20432ac2
Call-ID: 387ff28f-c1c7-401a-bdd9-b6e8055ef4c8
CSeq: 31214 ACK
Max-Forwards: 70
User-Agent: Asterisk PBX GIT-18-db824d8f6d
Content-Length:  0


Got  RTP packet from    192.168.1.65:9654 (type 00, seq 003420, ts 1069035691, len 000320)
Sent RTP packet to      192.168.1.51:4006 (type 00, seq 022712, ts 1069035688, len 000320)
<--- Received SIP response (983 bytes) from UDP:192.168.1.51:46917 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.59:5060;rport=5060;received=192.168.1.59;branch=z9hG4bKPjd4a10d87-e8b8-43df-aeda-43762a01f5a3
Call-ID: 387ff28f-c1c7-401a-bdd9-b6e8055ef4c8
From: "VideoCitofono" <sip:VideoCitofono@192.168.1.59>;tag=934f19ed-9993-456a-ad90-0a4d2f66e71a
To: <sip:1@192.168.1.51;ob>;tag=87036ac9-5686-4602-b003-431c20432ac2
CSeq: 31214 INVITE
Session-Expires: 1800;refresher=uac
Require: timer
Contact: <sip:1@192.168.1.51:46917;ob>
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Supported: replaces, 100rel, timer, norefersub
Content-Type: application/sdp
Content-Length:   316

v=0
o=- 3862623641 3862623643 IN IP4 192.168.1.51
s=pjmedia
b=AS:117
t=0 0
a=X-nat:0
m=audio 4006 RTP/AVP 0 101
c=IN IP4 192.168.1.51
b=TIAS:96000
a=rtcp:4007 IN IP4 192.168.1.51
a=sendrecv
a=rtpmap:0 PCMU/8000
a=ssrc:761689561 cname:3acf37564e7f7c23
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16

<--- Transmitting SIP request (451 bytes) to UDP:192.168.1.51:46917 --->
ACK sip:1@192.168.1.51:46917;ob SIP/2.0
Via: SIP/2.0/UDP 192.168.1.59:5060;rport;branch=z9hG4bKPjb37d0052-bbcb-421e-82c0-589f3c2302a6
From: "VideoCitofono" <sip:VideoCitofono@192.168.1.59>;tag=934f19ed-9993-456a-ad90-0a4d2f66e71a
To: <sip:1@192.168.1.51;ob>;tag=87036ac9-5686-4602-b003-431c20432ac2
Call-ID: 387ff28f-c1c7-401a-bdd9-b6e8055ef4c8
CSeq: 31214 ACK
Max-Forwards: 70
User-Agent: Asterisk PBX GIT-18-db824d8f6d
Content-Length:  0

RTP_debugAsterisk-18.txt

marcello-S551LB*CLI> rtp set debug on
RTP Packet Debugging Enabled
    -- Added contact 'sip:1@192.168.1.51:46917;ob' to AOR '1' with expiration of 300 seconds
  == Endpoint 1 is now Reachable
    -- Executing [1@internal:1] Dial("PJSIP/VideoCitofono-00000000", "PJSIP/1,60") in new stack
    -- Called PJSIP/1
    -- PJSIP/1-00000001 is ringing
    -- PJSIP/1-00000001 answered PJSIP/VideoCitofono-00000000
    -- Channel PJSIP/1-00000001 joined 'simple_bridge' basic-bridge <ab10becd-7c15-478c-a55f-1403c61758d2>
    -- Channel PJSIP/VideoCitofono-00000000 joined 'simple_bridge' basic-bridge <ab10becd-7c15-478c-a55f-1403c61758d2>
Got  RTP packet from    192.168.1.65:9654 (type 00, seq 004429, ts 1048417430, len 000320)
Sent RTP packet to      192.168.1.51:4000 (type 00, seq 019100, ts 1048417424, len 000320)
Got  RTP packet from    192.168.1.65:9654 (type 00, seq 004430, ts 1048417749, len 000320)
Sent RTP packet to      192.168.1.51:4000 (type 00, seq 019101, ts 1048417744, len 000320)
Got  RTP packet from    192.168.1.65:9654 (type 00, seq 004431, ts 1048418069, len 000320)
Sent RTP packet to      192.168.1.51:4000 (type 00, seq 019102, ts 1048418064, len 000320)
    -- Channel PJSIP/1-00000001 left 'native_rtp' basic-bridge <ab10becd-7c15-478c-a55f-1403c61758d2>
    -- Channel PJSIP/VideoCitofono-00000000 left 'native_rtp' basic-bridge <ab10becd-7c15-478c-a55f-1403c61758d2>
  == Spawn extension (internal, 1, 1) exited non-zero on 'PJSIP/VideoCitofono-00000000'

VC_asterisk_debug(Call-_Respone).txt

Connected to Asterisk GIT-18-db824d8f6d currently running on marcello-S551LB (pid = 1360)
marcello-S551LB*CLI> pjsip set logger host 192.168.1.65
PJSIP Logging Enabled for host: 192.168.1.65
<--- Received SIP request (747 bytes) from UDP:192.168.1.65:5060 --->
INVITE sip:1@192.168.1.59:5060 SIP/2.0
Max-Forwards: 20
Via: SIP/2.0/UDP 192.168.1.65:5060;rport;branch=z9hG4bK352258782
From: "VideoCitofono" <sip:VideoCitofono@192.168.1.65>;tag=965620315
To: <sip:1@192.168.1.59:5060>
Call-ID: 1688103885@192.168.1.65
CSeq: 77 INVITE
User-Agent: YATE/5.5.0
Contact: <sip:VideoCitofono@192.168.1.65:5060>
Allow: ACK, INVITE, BYE, CANCEL, MESSAGE, OPTIONS, INFO
Content-Type: application/sdp
Content-Length: 288

v=0
o=yate 1653664376 1653664376 IN IP4 192.168.1.65
s=SIP Call
c=IN IP4 192.168.1.65
t=0 0
m=audio 9654 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
m=video 9856 RTP/AVP 96
a=rtpmap:96 H264/90000
a=fmtp:96 profile-level-id=4D001F; packetization-mode=1

<--- Transmitting SIP response (495 bytes) to UDP:192.168.1.65:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.65:5060;rport=5060;received=192.168.1.65;branch=z9hG4bK352258782
Call-ID: 1688103885@192.168.1.65
From: "VideoCitofono" <sip:VideoCitofono@192.168.1.65>;tag=965620315
To: <sip:1@192.168.1.59>;tag=z9hG4bK352258782
CSeq: 77 INVITE
WWW-Authenticate: Digest realm="asterisk",nonce="1653635575/ee91f0e1f05e937874c846ae0819828b",opaque="288758af41df3c44",algorithm=MD5,qop="auth"
Server: Asterisk PBX GIT-18-db824d8f6d
Content-Length:  0


<--- Received SIP request (384 bytes) from UDP:192.168.1.65:5060 --->
ACK sip:1@192.168.1.59:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.65:5060;rport;branch=z9hG4bK352258782
From: "VideoCitofono" <sip:VideoCitofono@192.168.1.65>;tag=965620315
To: <sip:1@192.168.1.59:5060>;tag=z9hG4bK352258782
Call-ID: 1688103885@192.168.1.65
CSeq: 77 ACK
Max-Forwards: 20
Contact: <sip:VideoCitofono@192.168.1.65:5060>
User-Agent: YATE/5.5.0
Content-Length: 0


<--- Received SIP request (1051 bytes) from UDP:192.168.1.65:5060 --->
INVITE sip:1@192.168.1.59:5060 SIP/2.0
Max-Forwards: 20
Via: SIP/2.0/UDP 192.168.1.65:5060;rport;branch=z9hG4bK1745164313
From: "VideoCitofono" <sip:VideoCitofono@192.168.1.65>;tag=965620315
To: <sip:1@192.168.1.59:5060>
Call-ID: 1688103885@192.168.1.65
User-Agent: YATE/5.5.0
Contact: <sip:VideoCitofono@192.168.1.65:5060>
Allow: ACK, INVITE, BYE, CANCEL, MESSAGE, OPTIONS, INFO
CSeq: 78 INVITE
Authorization: Digest username="VideoCitofono", realm="asterisk", nonce="1653635575/ee91f0e1f05e937874c846ae0819828b", uri="sip:1@192.168.1.59:5060", response="250246a9fd24597349d0770abd932a81", algorithm=MD5, opaque="288758af41df3c44", qop=auth, nc=00000025, cnonce="1c8765bbb35c89db4668275d56612e11"
Content-Type: application/sdp
Content-Length: 288

v=0
o=yate 1653664376 1653664376 IN IP4 192.168.1.65
s=SIP Call
c=IN IP4 192.168.1.65
t=0 0
m=audio 9654 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
m=video 9856 RTP/AVP 96
a=rtpmap:96 H264/90000
a=fmtp:96 profile-level-id=4D001F; packetization-mode=1

<--- Transmitting SIP response (323 bytes) to UDP:192.168.1.65:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.65:5060;rport=5060;received=192.168.1.65;branch=z9hG4bK1745164313
Call-ID: 1688103885@192.168.1.65
From: "VideoCitofono" <sip:VideoCitofono@192.168.1.65>;tag=965620315
To: <sip:1@192.168.1.59>
CSeq: 78 INVITE
Server: Asterisk PBX GIT-18-db824d8f6d
Content-Length:  0


    -- Executing [1@internal:1] Dial("PJSIP/VideoCitofono-00000008", "PJSIP/1,60") in new stack
    -- Called PJSIP/1
    -- PJSIP/1-00000009 is ringing
    -- PJSIP/1-00000009 is ringing
<--- Transmitting SIP response (510 bytes) to UDP:192.168.1.65:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.1.65:5060;rport=5060;received=192.168.1.65;branch=z9hG4bK1745164313
Call-ID: 1688103885@192.168.1.65
From: "VideoCitofono" <sip:VideoCitofono@192.168.1.65>;tag=965620315
To: <sip:1@192.168.1.59>;tag=9bc80565-8d0b-4fad-98a5-b2acb853a770
CSeq: 78 INVITE
Server: Asterisk PBX GIT-18-db824d8f6d
Contact: <sip:192.168.1.59:5060>
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Content-Length:  0


    -- PJSIP/1-00000009 is ringing
    -- PJSIP/1-00000009 answered PJSIP/VideoCitofono-00000008
<--- Transmitting SIP response (846 bytes) to UDP:192.168.1.65:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.65:5060;rport=5060;received=192.168.1.65;branch=z9hG4bK1745164313
Call-ID: 1688103885@192.168.1.65
From: "VideoCitofono" <sip:VideoCitofono@192.168.1.65>;tag=965620315
To: <sip:1@192.168.1.59>;tag=9bc80565-8d0b-4fad-98a5-b2acb853a770
CSeq: 78 INVITE
Server: Asterisk PBX GIT-18-db824d8f6d
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Contact: <sip:192.168.1.59:5060>
Supported: 100rel, timer, replaces, norefersub
Content-Type: application/sdp
Content-Length:   259

v=0
o=- 1653664376 1653664378 IN IP4 192.168.1.59
s=Asterisk
c=IN IP4 192.168.1.59
t=0 0
m=audio 16244 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:150
a=sendrecv
m=video 0 RTP/AVP 96

    -- Channel PJSIP/1-00000009 joined 'simple_bridge' basic-bridge <c41c11dc-b674-4590-80a9-535e609084bb>
    -- Channel PJSIP/VideoCitofono-00000008 joined 'simple_bridge' basic-bridge <c41c11dc-b674-4590-80a9-535e609084bb>
Got  RTP packet from    192.168.1.65:9654 (type 00, seq 021813, ts 1074912903, len 000320)
Sent RTP packet to      192.168.1.51:4008 (type 00, seq 027049, ts 1074912896, len 000320)
Got  RTP packet from    192.168.1.65:9654 (type 00, seq 021814, ts 1074913222, len 000320)
Sent RTP packet to      192.168.1.51:4008 (type 00, seq 027050, ts 1074913216, len 000320)
<--- Received SIP request (703 bytes) from UDP:192.168.1.65:5060 --->
ACK sip:192.168.1.59:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.65:5060;rport;branch=z9hG4bK827219070
From: "VideoCitofono" <sip:VideoCitofono@192.168.1.65>;tag=965620315
To: <sip:192.168.1.59:5060>;tag=9bc80565-8d0b-4fad-98a5-b2acb853a770
Call-ID: 1688103885@192.168.1.65
CSeq: 78 ACK
Max-Forwards: 20
Contact: <sip:VideoCitofono@192.168.1.65:5060>
Authorization: Digest username="VideoCitofono", realm="asterisk", nonce="1653635575/ee91f0e1f05e937874c846ae0819828b", uri="sip:1@192.168.1.59:5060", response="250246a9fd24597349d0770abd932a81", algorithm=MD5, opaque="288758af41df3c44", qop=auth, nc=00000025, cnonce="1c8765bbb35c89db4668275d56612e11"
User-Agent: YATE/5.5.0
Content-Length: 0


<--- Transmitting SIP request (950 bytes) to UDP:192.168.1.65:5060 --->
INVITE sip:VideoCitofono@192.168.1.65:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.59:5060;rport;branch=z9hG4bKPjeec909e0-35e4-4c0f-bcfb-3336e89885d3
From: <sip:1@192.168.1.59>;tag=9bc80565-8d0b-4fad-98a5-b2acb853a770
To: "VideoCitofono" <sip:VideoCitofono@192.168.1.65>;tag=965620315
Contact: <sip:192.168.1.59:5060>
Call-ID: 1688103885@192.168.1.65
CSeq: 115 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub, histinfo
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: Asterisk PBX GIT-18-db824d8f6d
Content-Type: application/sdp
Content-Length:   258

v=0
o=- 1653664376 1653664379 IN IP4 192.168.1.59
s=Asterisk
c=IN IP4 192.168.1.51
t=0 0
m=audio 4008 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:150
a=sendrecv
m=video 0 RTP/AVP 96

<--- Received SIP response (372 bytes) from UDP:192.168.1.65:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.59:5060;rport=5060;branch=z9hG4bKPjeec909e0-35e4-4c0f-bcfb-3336e89885d3;received=192.168.1.59
From: <sip:1@192.168.1.59>;tag=9bc80565-8d0b-4fad-98a5-b2acb853a770
To: "VideoCitofono" <sip:VideoCitofono@192.168.1.65>;tag=965620315
Call-ID: 1688103885@192.168.1.65
CSeq: 115 INVITE
Server: YATE/5.5.0
Content-Length: 0


<--- Received SIP response (794 bytes) from UDP:192.168.1.65:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.59:5060;rport=5060;branch=z9hG4bKPjeec909e0-35e4-4c0f-bcfb-3336e89885d3;received=192.168.1.59
From: <sip:1@192.168.1.59>;tag=9bc80565-8d0b-4fad-98a5-b2acb853a770
To: "VideoCitofono" <sip:VideoCitofono@192.168.1.65>;tag=965620315
Call-ID: 1688103885@192.168.1.65
CSeq: 115 INVITE
Server: YATE/5.5.0
Contact: <sip:VideoCitofono@192.168.1.65:5060>
Allow: ACK, INVITE, BYE, CANCEL, MESSAGE, OPTIONS, INFO
Content-Type: application/sdp
Content-Length: 288

v=0
o=yate 1653664376 1653664376 IN IP4 192.168.1.65
s=SIP Call
c=IN IP4 192.168.1.65
t=0 0
m=audio 9654 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
m=video 9856 RTP/AVP 96
a=rtpmap:96 H264/90000
a=fmtp:96 profile-level-id=4D001F; packetization-mode=1

<--- Transmitting SIP request (414 bytes) to UDP:192.168.1.65:5060 --->
ACK sip:VideoCitofono@192.168.1.65:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.59:5060;rport;branch=z9hG4bKPjc7473117-a77b-486d-bc6e-1666c800d710
From: <sip:1@192.168.1.59>;tag=9bc80565-8d0b-4fad-98a5-b2acb853a770
To: "VideoCitofono" <sip:VideoCitofono@192.168.1.65>;tag=965620315
Call-ID: 1688103885@192.168.1.65
CSeq: 115 ACK
Max-Forwards: 70
User-Agent: Asterisk PBX GIT-18-db824d8f6d
Content-Length:  0
2

You don’t have any allow lines for your video codec (H264, in this case).

3

thanks david,
I have place for each endpoint in “pjsip.conf” file
allow=H264
but it still not works.

I recive an error message from PJSUA2

pjsua_media.c .......Skipped
updating media call02:1 (media type=unknown): Unsupported media type
(PJMEDIA_EUNSUPMEDIATYPE)

Where can my mistake be?
thanks

4

Are you using a really ancient PJSUA, in which case see:

https://trac.pjsip.org/repos/changeset/1689

I also noticed in your original trace that Asterisk said: do not use video, but PJSUA made an offer of video, which looks wrong to me.

5

tks David,
I’m using the latest PJSUA2 version, but I think that the problem is not in PJSUA2, because the program works with other Sip Server. I belive that some Asterisk parameters (configuration) should be setted but I don’t know witchone.

Moreover when I follow your suggestion (many thanks) to use “allow = ulaw,h264” in any endpoit, something good happens: I can hear some sounds (even with Larsen effect since my PC and the video intercom are near).
BUT after a few seconds (3 sec more or less) sound stops and the following message appears:


`20:52:19.265           sound_port.c  EC suspended because of inactivity`

thanks for your help

6

As a pedantic point, this issue can only arise when it is interacting with a SIP client. Asterisk acts as both.

7

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