asterisk.ru

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  == Spawn extension (factory, 101, 1) exited non-zero on 'SIP/102-0000000a'
  == Using SIP VIDEO CoS mark 6
  == Using SIP RTP CoS mark 5
    -- Executing [101@factory:1] Dial("SIP/102-0000000c", "SIP/101")
  == Using SIP VIDEO CoS mark 6
  == Using SIP RTP CoS mark 5
    -- Called SIP/101
    -- SIP/101-0000000d is ringing
    -- SIP/101-0000000d answered SIP/102-0000000c
    -- Remotely bridging SIP/102-0000000c and SIP/101-0000000d
       > 0xb6c20930 -- Probation passed - setting RTP source address to 192.168.111.35:7078
       > 0xb6b426f8 -- Probation passed - setting RTP source address to 192.168.111.33:62700
       > 0xb6b269c0 -- Probation passed - setting RTP source address to 192.168.111.33:62110
[Sep 10 17:52:41] NOTICE[3647][C-00000006]: res_rtp_asterisk.c:3767 ast_rtp_read: Unknown RTP codec 125 received from '192.168.111.33:62110'

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<--- SIP read from UDP:192.168.111.33:37692 --->
INVITE sip:104@192.168.111.41 SIP/2.0
Via: SIP/2.0/UDP 192.168.111.33:37692;branch=z9hG4bK-524287-1---1c263b38368b2411;rport
Max-Forwards: 70
Contact: <sip:101@192.168.111.33:37692>;+sip.instance="<urn:uuid:de194b19f5af1f482fb63bde20aed10f19de84be>"
To: <sip:104@192.168.111.41>
From: "tesrik"<sip:101@192.168.111.41>;tag=7674817f
Call-ID: 126264M2NkZDA1MDExZjVlZjlhM2U0YmRiZWMwOTdkYWNmYzg
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, BYE, REFER, INFO, NOTIFY, OPTIONS, SUBSCRIBE, MESSAGE
Content-Type: application/sdp
Supported: outbound, path
User-Agent: Bria Android release 3.4.1 stamp 81288
Content-Length: 388

v=0
o=- 617990633729 1 IN IP4 192.168.111.33
s=Cpc session
c=IN IP4 192.168.111.33
t=0 0
m=audio 52030 RTP/AVP 0 102
a=rtpmap:102 ILBC/8000
a=sendrecv
m=video 55288 RTP/AVP 126 127
a=rtpmap:126 H264/90000
a=fmtp:126 profile-level-id=428016;packetization-mode=1
a=rtpmap:127 H264/90000
a=fmtp:127 profile-level-id=428016;packetization-mode=0
a=rtcp-fb:* nack pli
a=sendrecv
<------------->
--- (13 headers 15 lines) ---
Sending to 192.168.111.33:37692 (no NAT)
Sending to 192.168.111.33:37692 (no NAT)
Using INVITE request as basis request - 126264M2NkZDA1MDExZjVlZjlhM2U0YmRiZWMwOTdkYWNmYzg
Scheduling destruction of SIP dialog '589d2803258e9f1337b1e96957dcfe4f@192.168.111.41' in 32000 ms (Method: NOTIFY)
Reliably Transmitting (no NAT) to 192.168.111.33:37692:
NOTIFY sip:101@192.168.111.33:37692 SIP/2.0
Via: SIP/2.0/UDP 192.168.111.41:5060;branch=z9hG4bK661e7490
Max-Forwards: 70
From: "asterisk" <sip:101@192.168.111.41>;tag=as360a380f
To: <sip:101@192.168.111.33:37692>
Contact: <sip:101@192.168.111.41:5060>
Call-ID: 589d2803258e9f1337b1e96957dcfe4f@192.168.111.41
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX 11.7.0~dfsg-1ubuntu1
Event: message-summary
Content-Type: application/simple-message-summary
Content-Length: 94

Messages-Waiting: no
Message-Account: sip:asterisk@192.168.111.41
Voice-Message: 0/0 (0/0)

---
Found peer '101' for '101' from 192.168.111.33:37692
  == Using SIP VIDEO CoS mark 6
  == Using SIP RTP CoS mark 5
Found RTP audio format 0
Found RTP audio format 102
Found audio description format ILBC for ID 102
Found RTP video format 126
Found RTP video format 127
Found video description format H264 for ID 126
Found video description format H264 for ID 127
Capabilities: us - (gsm|ulaw|alaw|h263|h264|testlaw), peer - audio=(ulaw|ilbc)/video=(h264)/text=(nothing), combined - (ulaw|h264)
Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x0 (nothing), combined - 0x0 (nothing)
Peer audio RTP is at port 192.168.111.33:52030
Peer video RTP is at port 192.168.111.33:55288
Looking for 104 in factory (domain 192.168.111.41)
list_route: hop: <sip:101@192.168.111.33:37692>

Код: Выделить всё

m=video 55288 RTP/AVP 126 127
a=rtpmap:126 H264/90000
a=fmtp:126 profile-level-id=428016;packetization-mode=1
a=rtpmap:127 H264/90000
a=fmtp:127 profile-level-id=428016;packetization-mode=0