[Gelöst] SIP DEBUG eingehenden Anruf T-COM DeutschlandLAN IP Voice/ Data S

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vwittich

vwittich

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Ich stolpere (kein Audio) mal wieder über die T-Com. Diesmal an einem Anschluss mit DeutschlandLAN IP Voice/ Data S und fester IP.

Der SIP DEBUG meldet in meinen Auge nichts verdächtiges:

Code: 
<--- SIP read from UDP:217.0.23.39:5060 --->
INVITE sip:[email protected]:5060 SIP/2.0
Max-Forwards: 63
Via: SIP/2.0/UDP 217.0.23.39:5060;branch=Ohgh2oog3Zqkv7ig6yho1irnv7co5sgy5ngtbhl7
To: <sip:[email protected];user=phone>
From: <sip:[email protected];user=phone>;tag=h7g4Esbg_p65550t1454705141m950005c67912034s1_4013769360-1780043889
Call-ID: p65550t1454705141m950005c67912034s2
CSeq: 1 INVITE
Contact: <sip:[email protected];transport=udp>;+g.3gpp.icsi-ref="urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel"
Record-Route: <sip:217.0.23.39;transport=udp;lr>
Accept-Contact: *;+g.3gpp.icsi-ref="urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel"
Min-Se: 900
P-Asserted-Identity: <sip:[email protected];user=phone>
Privacy: none
Session-Expires: 1800
Supported: timer
Supported: histinfo
Content-Type: application/sdp
Content-Length: 339
Session-ID: da30d534cf38cc3cf5249d279a2a9f2d
Allow: REGISTER, REFER, NOTIFY, SUBSCRIBE, INFO, PRACK, UPDATE, INVITE, ACK, OPTIONS, CANCEL, BYE

v=0
o=- 1443686064 4013768858 IN IP4 217.0.23.39
s=-
c=IN IP4 217.0.4.164
t=0 0
m=audio 51294 RTP/AVP 8 0 18 2 38 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:2 G726-32/8000
a=rtpmap:38 G726-40/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
<------------->
[Feb  5 21:45:42] VERBOSE[10457] chan_sip.c: --- (20 headers 16 lines) ---
[Feb  5 21:45:42] VERBOSE[10457] chan_sip.c: Sending to 217.0.23.39:5060 (NAT)
[Feb  5 21:45:42] VERBOSE[10457] chan_sip.c: Using INVITE request as basis request - p655...s2
[Feb  5 21:45:42] VERBOSE[10457] chan_sip.c: Found peer 'DTAG-IP_IN23_39' for '+4969333333333' from 217.0.23.39:5060
[Feb  5 21:45:42] VERBOSE[10457] netsock2.c:   == Using SIP RTP CoS mark 5
[Feb  5 21:45:42] VERBOSE[10457] chan_sip.c: Found RTP audio format 8
[Feb  5 21:45:42] VERBOSE[10457] chan_sip.c: Found RTP audio format 0
[Feb  5 21:45:42] VERBOSE[10457] chan_sip.c: Found RTP audio format 18
[Feb  5 21:45:42] VERBOSE[10457] chan_sip.c: Found RTP audio format 2
[Feb  5 21:45:42] VERBOSE[10457] chan_sip.c: Found RTP audio format 38
[Feb  5 21:45:42] VERBOSE[10457] chan_sip.c: Found RTP audio format 101
[Feb  5 21:45:42] VERBOSE[10457] chan_sip.c: Found audio description format PCMA for ID 8
[Feb  5 21:45:42] VERBOSE[10457] chan_sip.c: Found audio description format PCMU for ID 0
[Feb  5 21:45:42] VERBOSE[10457] chan_sip.c: Found audio description format G729 for ID 18
[Feb  5 21:45:42] VERBOSE[10457] chan_sip.c: Found audio description format G726-32 for ID 2
[Feb  5 21:45:42] VERBOSE[10457] chan_sip.c: Found unknown media description format G726-40 for ID 38
[Feb  5 21:45:42] VERBOSE[10457] chan_sip.c: Found audio description format telephone-event for ID 101
[Feb  5 21:45:42] VERBOSE[10457] chan_sip.c: Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x90c (ulaw|alaw|g726|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw)
[Feb  5 21:45:42] VERBOSE[10457] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
[Feb  5 21:45:42] VERBOSE[10457] chan_sip.c: Peer audio RTP is at port 217.0.4.164:51294
[Feb  5 21:45:42] VERBOSE[10457] chan_sip.c: Looking for 06955555555 in von_telekom (domain 80.147.200.001)
[Feb  5 21:45:42] VERBOSE[10457] chan_sip.c: list_route: hop: <sip:217.0.23.39;transport=udp;lr>
[Feb  5 21:45:42] VERBOSE[10457] chan_sip.c: 
<--- Transmitting (NAT) to 217.0.23.39:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 217.0.23.39:5060;branch=Ohgh2oog3Zqkv7ig6yho1irnv7co5sgy5ngtbhl7;received=217.0.23.39;rport=5060
Record-Route: <sip:217.0.23.39;transport=udp;lr>
From: <sip:[email protected];user=phone>;tag=h7g4Esbg_p65550t1454705141m950005c67912034s1_4013769360-1780043889
To: <sip:[email protected];user=phone>
Call-ID: p65550t1454705141m950005c67912034s2
CSeq: 1 INVITE
Server: Asterisk PBX 1.8.13.1~dfsg1-3+deb7u3
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces
Contact: <sip:[email protected]:5060>
Content-Length: 0


<------------>
[Feb  5 21:45:42] VERBOSE[32601] pbx.c:     -- Executing [06955555555@von_telekom:1] NoOp("SIP/DTAG-IP_IN23_39-00000b72", "Testcall via Telekom") in new stack
[Feb  5 21:45:42] VERBOSE[32601] pbx.c:     -- Executing [06955555555@von_telekom:2] Ringing("SIP/DTAG-IP_IN23_39-00000b72", "") in new stack
[Feb  5 21:45:42] VERBOSE[32601] chan_sip.c: 
<--- Transmitting (NAT) to 217.0.23.39:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 217.0.23.39:5060;branch=Ohgh2oog3Zqkv7ig6yho1irnv7co5sgy5ngtbhl7;received=217.0.23.39;rport=5060
Record-Route: <sip:217.0.23.39;transport=udp;lr>
From: <sip:[email protected];user=phone>;tag=h7g4Esbg_p65550t1454705141m950005c67912034s1_4013769360-1780043889
To: <sip:[email protected];user=phone>;tag=as32422ed3
Call-ID: p65550t1454705141m950005c67912034s2
CSeq: 1 INVITE
Server: Asterisk PBX 1.8.13.1~dfsg1-3+deb7u3
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces
Contact: <sip:[email protected]:5060>
Content-Length: 0


<------------>
[Feb  5 21:45:42] VERBOSE[32601] pbx.c:     -- Executing [06955555555@von_telekom:3] Wait("SIP/DTAG-IP_IN23_39-00000b72", "3") in new stack
[Feb  5 21:45:45] VERBOSE[32601] pbx.c:     -- Executing [06955555555@von_telekom:4] Answer("SIP/DTAG-IP_IN23_39-00000b72", "") in new stack
[Feb  5 21:45:45] VERBOSE[32601] chan_sip.c: Audio is at 13472
[Feb  5 21:45:45] VERBOSE[32601] chan_sip.c: Adding codec 0x8 (alaw) to SDP
[Feb  5 21:45:45] VERBOSE[32601] chan_sip.c: Adding codec 0x4 (ulaw) to SDP
[Feb  5 21:45:45] VERBOSE[32601] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
[Feb  5 21:45:45] VERBOSE[32601] chan_sip.c: 
<--- Reliably Transmitting (NAT) to 217.0.23.39:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 217.0.23.39:5060;branch=Ohgh2oog3Zqkv7ig6yho1irnv7co5sgy5ngtbhl7;received=217.0.23.39;rport=5060
Record-Route: <sip:217.0.23.39;transport=udp;lr>
From: <sip:[email protected];user=phone>;tag=h7g4Esbg_p65550t1454705141m950005c67912034s1_4013769360-1780043889
To: <sip:[email protected];user=phone>;tag=as32422ed3
Call-ID: p65550t1454705141m950005c67912034s2
CSeq: 1 INVITE
Server: Asterisk PBX 1.8.13.1~dfsg1-3+deb7u3
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces
Contact: <sip:[email protected]:5060>
Content-Type: application/sdp
Content-Length: 278

v=0
o=root 737623202 737623202 IN IP4 80.147.200.001
s=Asterisk PBX 1.8.13.1~dfsg1-3+deb7u3
c=IN IP4 80.147.200.001
t=0 0
m=audio 13472 RTP/AVP 8 0 101
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=sendrecv

<------------>
[Feb  5 21:45:45] VERBOSE[10457] chan_sip.c: 
<--- SIP read from UDP:217.0.23.39:5060 --->
ACK sip:[email protected]:5060 SIP/2.0
Max-Forwards: 65
Via: SIP/2.0/UDP 217.0.23.39:5060;branch=Ohgh2oog3Zqkv7ikb12o6g1xt9r2peg5byjg5s9w
To: <sip:[email protected];user=phone>;tag=as32422ed3
From: <sip:[email protected];user=phone>;tag=h7g4Esbg_p65550t1454705141m950005c67912034s1_4013769360-1780043889
Call-ID: p65550t1454705141m950005c67912034s2
CSeq: 1 ACK
Contact: <sip:[email protected];transport=udp>;+g.3gpp.icsi-ref="urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel"
Content-Length: 0

<------------->
[Feb  5 21:45:45] VERBOSE[10457] chan_sip.c: --- (9 headers 0 lines) ---
[Feb  5 21:45:45] VERBOSE[32601] pbx.c:     -- Executing [06955555555@von_telekom:5] Playback("SIP/DTAG-IP_IN23_39-00000b72", "tt-weasels") in new stack
[Feb  5 21:45:45] VERBOSE[32601] file.c:     -- <SIP/DTAG-IP_IN23_39-00000b72> Playing 'tt-weasels.gsm' (language 'de')
[Feb  5 21:45:48] VERBOSE[32601] pbx.c:     -- Executing [06955555555@von_telekom:6] Wait("SIP/DTAG-IP_IN23_39-00000b72", "1") in new stack
[Feb  5 21:45:49] VERBOSE[32601] pbx.c:     -- Executing [06955555555@von_telekom:7] NoOp("SIP/DTAG-IP_IN23_39-00000b72", "Going to hangup Testcall") in new stack
[Feb  5 21:45:49] VERBOSE[32601] pbx.c:     -- Executing [06955555555@von_telekom:8] Hangup("SIP/DTAG-IP_IN23_39-00000b72", "") in new stack
[Feb  5 21:45:49] VERBOSE[32601] pbx.c:   == Spawn extension (von_telekom, 06955555555, 8) exited non-zero on 'SIP/DTAG-IP_IN23_39-00000b72'
[Feb  5 21:45:49] VERBOSE[32601] chan_sip.c: Scheduling destruction of SIP dialog 'p65550t1454705141m950005c67912034s2' in 32000 ms (Method: ACK)
[Feb  5 21:45:49] VERBOSE[32601] chan_sip.c: set_destination: Parsing <sip:217.0.23.39;transport=udp;lr> for address/port to send to
[Feb  5 21:45:49] VERBOSE[32601] chan_sip.c: set_destination: set destination to 217.0.23.39:5060
[Feb  5 21:45:49] VERBOSE[32601] chan_sip.c: Reliably Transmitting (NAT) to 217.0.23.39:5060:
BYE sip:[email protected];transport=udp SIP/2.0
Via: SIP/2.0/UDP 80.147.200.001:5060;branch=Ohgh2oo74c789a3;rport
Route: <sip:217.0.23.39;transport=udp;lr>
Max-Forwards: 70
From: <sip:[email protected];user=phone>;tag=as32422ed3
To: <sip:[email protected];user=phone>;tag=h7g4Esbg_p65550t1454705141m950005c67912034s1_4013769360-1780043889
Call-ID: p65550t1454705141m950005c67912034s2
CSeq: 102 BYE
User-Agent: Asterisk PBX 1.8.13.1~dfsg1-3+deb7u3
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


---
[Feb  5 21:45:49] VERBOSE[10457] chan_sip.c: 
<--- SIP read from UDP:217.0.23.39:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 80.147.200.001:5060;received=80.147.200.001;rport=38148;branch=Ohgh2oo74c789a3
To: <sip:[email protected];user=phone>;tag=h7g4Esbg_p65550t1454705141m950005c67912034s1_4013769360-1780043889
From: <sip:[email protected];user=phone>;tag=as32422ed3
Call-ID: p65550t1454705141m950005c67912034s2
CSeq: 102 BYE
Content-Length: 0
Allow: REGISTER, REFER, NOTIFY, SUBSCRIBE, INFO, PRACK, UPDATE, INVITE, ACK, OPTIONS, CANCEL, BYE

Zusätzlich habe ich mir die den RTP DEBUG angeschaut:
Code: 
 rtp set debug on
RTP Debugging Enabled
  == Using SIP RTP CoS mark 5
    -- Executing [06955555555@von_telekom:1] NoOp("SIP/DTAG-IP_IN23_39-00000b81", "Testcall via Telekom") in new stack
    -- Executing [06955555555@von_telekom:2] Ringing("SIP/DTAG-IP_IN23_39-00000b81", "") in new stack
    -- Executing [06955555555@von_telekom:3] Wait("SIP/DTAG-IP_IN23_39-00000b81", "3") in new stack
    -- Executing [06955555555@von_telekom:4] Answer("SIP/DTAG-IP_IN23_39-00000b81", "") in new stack
    -- Executing [06955555555@von_telekom:5] Playback("SIP/DTAG-IP_IN23_39-00000b81", "tt-weasels") in new stack
Sent RTP packet to      217.0.5.100:57260 (type 08, seq 040154, ts 000160, len 000160)
    -- <SIP/DTAG-IP_IN23_39-00000b81> Playing 'tt-weasels.gsm' (language 'de')
Sent RTP packet to      217.0.5.100:57260 (type 08, seq 040155, ts 000320, len 000160)
Sent RTP packet to      217.0.5.100:57260 (type 08, seq 040156, ts 000480, len 000160)
Sent RTP packet to      217.0.5.100:57260 (type 08, seq 040157, ts 000640, len 000160)
[...]
Sent RTP packet to      217.0.5.100:57260 (type 08, seq 040286, ts 021280, len 000160)
Sent RTP packet to      217.0.5.100:57260 (type 08, seq 040287, ts 021440, len 000160)
    -- Executing [06955555555@von_telekom:6] Wait("SIP/DTAG-IP_IN23_39-00000b81", "1") in new stack
    -- Executing [06955555555@von_telekom:7] NoOp("SIP/DTAG-IP_IN23_39-00000b81", "Going to hangup Testcall") in new stack
    -- Executing [06955555555@von_telekom:8] Hangup("SIP/DTAG-IP_IN23_39-00000b81", "") in new stack
  == Spawn extension (von_telekom, 06955555555, 8) exited non-zero on 'SIP/DTAG-IP_IN23_39-00000b81'

Wenn ich den Anruf über Sipgate leite sehe ich im RTP DEBUG, dass die Pakete nicht nur gesendet sonder auch empfangen werden. Als Resultat höre ich dann auch das Audio-Signal.
Code: 
Sent RTP packet to      217.10.77.241:29184 (type 08, seq 030684, ts 000320, len 000160)
Sent RTP packet to      217.10.77.241:29184 (type 08, seq 030685, ts 000480, len 000160)
Got  RTP packet from    217.10.77.241:29184 (type 08, seq 011289, ts 1777121518, len 000160)
Sent RTP packet to      217.10.77.241:29184 (type 08, seq 030686, ts 000640, len 000160)
Got  RTP packet from    217.10.77.241:29184 (type 08, seq 011290, ts 1777121678, len 000160)
Sent RTP packet to      217.10.77.241:29184 (type 08, seq 030687, ts 000800, len 000160)
Got  RTP packet from    217.10.77.241:29184 (type 08, seq 011291, ts 1777121838, len 000160)
Sent RTP packet to      217.10.77.241:29184 (type 08, seq 030688, ts 000960, len 000160)
Got  RTP packet from    217.10.77.241:29184 (type 08, seq 011292, ts 1777121998, len 000160)
Sent RTP packet to      217.10.77.241:29184 (type 08, seq 030689, ts 001120, len 000160)
Got  RTP packet from    217.10.77.241:29184 (type 08, seq 011293, ts 1777122158, len 000160)

Wo liegt der Fehler? Warum werden die Audio-Pakete bei der eingehenden T-Com-Verbindung nicht ordnungsgemäß übertragen?
 
Zuletzt bearbeitet:
Lösung der fehlenden Audio Übertragung mit T-Com

Okay bin am nächsten Morgen mit klaren Gedanken der Sache auf die Spur gekommen.

Scheinbar ist Telekom sehr sensible was die RTP Ports angeht. Also die Standard-Ports in /etc/asterisk/rtp.conf auf 30000 - 31000 gestellt (diese waren im Router bereits freigegeben).

Code: 
[general]
rtpstart=30000
rtpend=31000
 
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