hello,
here is the sip trace with rtp, rtcp debuging. i make a call to 444 (beep-tone) with following record-function.
217.92.X.X → Public IP of my Homeoffice Router.
217.7.X.X → Public IP of the Asterisk
[code]<— SIP read from UDP:217.92.X.X:2048 —>
INVITE sip:444@217.7.X.X SIP/2.0
Via: SIP/2.0/UDP 217.92.X.X:2048;branch=z9hG4bK-dsys34jbftzm;rport
From: sip:user@217.7.X.X;tag=kf0zzs6jky
To: sip:444@217.7.X.X
Call-ID: 3c27a0e8ce1d-6kwv55572kdx
CSeq: 1 INVITE
Max-Forwards: 70
Contact: sip:user@217.92.X.X:2048;reg-id=1
X-Serialnumber: 000413383485
P-Key-Flags: keys=“3”
User-Agent: snom320/8.4.32
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE
Allow-Events: talk, hold, refer, call-info
Supported: timer, 100rel, replaces, from-change
Session-Expires: 3600;refresher=uas
Min-SE: 90
Content-Type: application/sdp
Content-Length: 477
v=0
o=root 1669376141 1669376141 IN IP4 217.92.X.X
s=call
c=IN IP4 217.92.X.X
t=0 0
m=audio 63024 RTP/AVP 0 8 9 99 3 18 4 101
a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:YgT6PUdoHKpoA7TVGPgrdbZEW0awM8FwHxcpZPCL
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=rtpmap:99 G726-32/8000
a=rtpmap:3 GSM/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:4 G723/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------->
— (19 headers 19 lines) —
Sending to 217.92.X.X:2048 (no NAT)
Using INVITE request as basis request - 3c27a0e8ce1d-6kwv55572kdx
Found peer ‘user’ for ‘user’ from 217.92.X.X:2048
<— Reliably Transmitting (NAT) to 217.92.X.X:2048 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 217.92.X.X:2048;branch=z9hG4bK-dsys34jbftzm;received=217.92.X.X;rport=2048
From: sip:user@217.7.X.X;tag=kf0zzs6jky
To: sip:444@217.7.X.X;tag=as62f32bc9
Call-ID: 3c27a0e8ce1d-6kwv55572kdx
CSeq: 1 INVITE
Server: Asterisk PBX 11.2.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce=“07a7ecd6”
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog ‘3c27a0e8ce1d-6kwv55572kdx’ in 6400 ms (Method: INVITE)
<— SIP read from UDP:217.92.X.X:2048 —>
ACK sip:444@217.7.X.X SIP/2.0
Via: SIP/2.0/UDP 217.92.X.X:2048;branch=z9hG4bK-dsys34jbftzm;rport
From: sip:user@217.7.X.X;tag=kf0zzs6jky
To: sip:444@217.7.X.X;tag=as62f32bc9
Call-ID: 3c27a0e8ce1d-6kwv55572kdx
CSeq: 1 ACK
Max-Forwards: 70
Contact: sip:user@217.92.X.X:2048;reg-id=1
Content-Length: 0
<------------->
— (9 headers 0 lines) —
<— SIP read from UDP:217.92.X.X:2048 —>
INVITE sip:444@217.7.X.X SIP/2.0
Via: SIP/2.0/UDP 217.92.X.X:2048;branch=z9hG4bK-s3084kknh7n0;rport
From: sip:user@217.7.X.X;tag=kf0zzs6jky
To: sip:444@217.7.X.X
Call-ID: 3c27a0e8ce1d-6kwv55572kdx
CSeq: 2 INVITE
Max-Forwards: 70
Contact: sip:user@217.92.X.X:2048;reg-id=1
X-Serialnumber: 000413383485
P-Key-Flags: keys=“3”
User-Agent: snom320/8.4.32
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE
Allow-Events: talk, hold, refer, call-info
Supported: timer, 100rel, replaces, from-change
Session-Expires: 3600;refresher=uas
Min-SE: 90
Authorization: Digest username=“user”,realm=“asterisk”,nonce=“07a7ecd6”,uri=“sip:444@217.7.X.X”,response=“47084ffc7e3ea362692a5e7ae26db3d9”,algorithm=MD5
Content-Type: application/sdp
Content-Length: 477
v=0
o=root 1669376141 1669376141 IN IP4 217.92.X.X
s=call
c=IN IP4 217.92.X.X
t=0 0
m=audio 63024 RTP/AVP 0 8 9 99 3 18 4 101
a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:YgT6PUdoHKpoA7TVGPgrdbZEW0awM8FwHxcpZPCL
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=rtpmap:99 G726-32/8000
a=rtpmap:3 GSM/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:4 G723/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------->
— (20 headers 19 lines) —
Sending to 217.92.X.X:2048 (NAT)
Using INVITE request as basis request - 3c27a0e8ce1d-6kwv55572kdx
Found peer ‘user’ for ‘user’ from 217.92.X.X:2048
== Using SIP RTP CoS mark 5
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 9
Found RTP audio format 99
Found RTP audio format 3
Found RTP audio format 18
Found RTP audio format 4
Found RTP audio format 101
[Apr 24 10:14:10] ERROR[2134][C-00000007]: chan_sip.c:32788 setup_srtp: No SRTP module loaded, can’t setup SRTP session.
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format G722 for ID 9
Found audio description format G726-32 for ID 99
Found audio description format GSM for ID 3
Found audio description format G729 for ID 18
Found audio description format G723 for ID 4
Found audio description format telephone-event for ID 101
Capabilities: us - (gsm|ulaw|alaw|g729), peer - audio=(g723|gsm|ulaw|alaw|g726|g729|g722)/video=(nothing)/text=(nothing), combined - (gsm|ulaw|alaw|g729)
Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x1 (telephone-event|), combined - 0x0 (nothing)
Peer audio RTP is at port 217.92.X.X:63024
Looking for 444 in it-administration (domain 217.7.X.X)
list_route: hop: sip:user@217.92.X.X:2048
<— Transmitting (NAT) to 217.92.X.X:2048 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 217.92.X.X:2048;branch=z9hG4bK-s3084kknh7n0;received=217.92.X.X;rport=2048
From: sip:user@217.7.X.X;tag=kf0zzs6jky
To: sip:444@217.7.X.X
Call-ID: 3c27a0e8ce1d-6kwv55572kdx
CSeq: 2 INVITE
Server: Asterisk PBX 11.2.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: sip:444@217.7.X.X:5060
Content-Length: 0
<------------>
– Executing [444@it-administration:1] Answer(“SIP/user-00000007”, “”) in new stack
Audio is at 20024
Adding codec 100003 (ulaw) to SDP
Adding codec 100004 (alaw) to SDP
Adding codec 100008 (g729) to SDP
Adding codec 100002 (gsm) to SDP
<— Reliably Transmitting (NAT) to 217.92.X.X:2048 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 217.92.X.X:2048;branch=z9hG4bK-s3084kknh7n0;received=217.92.X.X;rport=2048
From: sip:user@217.7.X.X;tag=kf0zzs6jky
To: sip:444@217.7.X.X;tag=as1d5e4c30
Call-ID: 3c27a0e8ce1d-6kwv55572kdx
CSeq: 2 INVITE
Server: Asterisk PBX 11.2.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: sip:444@217.7.X.X:5060
Content-Type: application/sdp
Require: timer
Content-Length: 271
v=0
o=root 471051302 471051302 IN IP4 217.7.X.X
s=Asterisk PBX 11.2.1
c=IN IP4 217.7.X.X
t=0 0
m=audio 20024 RTP/AVP 0 8 18 3
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:3 GSM/8000
a=ptime:20
a=sendrecv
<------------>
Retransmitting #1 (NAT) to 217.92.X.X:2048:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 217.92.X.X:2048;branch=z9hG4bK-s3084kknh7n0;received=217.92.X.X;rport=2048
From: sip:user@217.7.X.X;tag=kf0zzs6jky
To: sip:444@217.7.X.X;tag=as1d5e4c30
Call-ID: 3c27a0e8ce1d-6kwv55572kdx
CSeq: 2 INVITE
Server: Asterisk PBX 11.2.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: sip:444@217.7.X.X:5060
Content-Type: application/sdp
Require: timer
Content-Length: 271
v=0
o=root 471051302 471051302 IN IP4 217.7.X.X
s=Asterisk PBX 11.2.1
c=IN IP4 217.7.X.X
t=0 0
m=audio 20024 RTP/AVP 0 8 18 3
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:3 GSM/8000
a=ptime:20
a=sendrecv
<— SIP read from UDP:217.92.X.X:2048 —>
ACK sip:444@217.7.X.X:5060 SIP/2.0
Via: SIP/2.0/UDP 217.92.X.X:2048;branch=z9hG4bK-mkiuwx78fjvj;rport
From: sip:user@217.7.X.X;tag=kf0zzs6jky
To: sip:444@217.7.X.X;tag=as1d5e4c30
Call-ID: 3c27a0e8ce1d-6kwv55572kdx
CSeq: 2 ACK
Max-Forwards: 70
Contact: sip:user@217.92.X.X:2048;reg-id=1
Content-Length: 0
<------------->
— (9 headers 0 lines) —
<— SIP read from UDP:217.92.X.X:2048 —>
ACK sip:444@217.7.X.X:5060 SIP/2.0
Via: SIP/2.0/UDP 217.92.X.X:2048;branch=z9hG4bK-mkiuwx78fjvj;rport
From: sip:user@217.7.X.X;tag=kf0zzs6jky
To: sip:444@217.7.X.X;tag=as1d5e4c30
Call-ID: 3c27a0e8ce1d-6kwv55572kdx
CSeq: 2 ACK
Max-Forwards: 70
Contact: sip:user@217.92.X.X:2048;reg-id=1
Content-Length: 0
<------------->
— (9 headers 0 lines) —
– Executing [444@it-administration:2] Wait(“SIP/user-00000007”, “2”) in new stack
– Executing [444@it-administration:3] Record(“SIP/user-00000007”, “/drbd-asterisk/var/lib/asterisk/ivrs/technik/test.wav”) in new stack
Sent RTP packet to 217.92.X.X:63024 (type 00, seq 003684, ts 000160, len 000160)
– <SIP/user-00000007> Playing ‘beep.gsm’ (language ‘de’)
Sent RTP packet to 217.92.X.X:63024 (type 00, seq 003685, ts 000320, len 000160)
Sent RTP packet to 217.92.X.X:63024 (type 00, seq 003686, ts 000480, len 000160)
Sent RTP packet to 217.92.X.X:63024 (type 00, seq 003687, ts 000640, len 000160)
Sent RTP packet to 217.92.X.X:63024 (type 00, seq 003688, ts 000800, len 000160)
Sent RTP packet to 217.92.X.X:63024 (type 00, seq 003689, ts 000960, len 000160)
Sent RTP packet to 217.92.X.X:63024 (type 00, seq 003690, ts 001120, len 000160)
Sent RTP packet to 217.92.X.X:63024 (type 00, seq 003691, ts 001280, len 000160)
Sent RTP packet to 217.92.X.X:63024 (type 00, seq 003692, ts 001440, len 000160)
Sent RTP packet to 217.92.X.X:63024 (type 00, seq 003693, ts 001600, len 000160)
Sent RTP packet to 217.92.X.X:63024 (type 00, seq 003694, ts 001760, len 000160)
Sent RTP packet to 217.92.X.X:63024 (type 00, seq 003695, ts 001920, len 000160)
Sent RTP packet to 217.92.X.X:63024 (type 00, seq 003696, ts 002080, len 000160)
Sent RTP packet to 217.92.X.X:63024 (type 00, seq 003697, ts 002240, len 000160)
Sent RTP packet to 217.92.X.X:63024 (type 00, seq 003698, ts 002400, len 000160)
Sent RTP packet to 217.92.X.X:63024 (type 00, seq 003699, ts 002560, len 000160)
Sent RTP packet to 217.92.X.X:63024 (type 00, seq 003700, ts 002720, len 000160)
Sent RTP packet to 217.92.X.X:63024 (type 00, seq 003701, ts 002880, len 000160)
Sent RTP packet to 217.92.X.X:63024 (type 00, seq 003702, ts 003040, len 000160)
Sent RTP packet to 217.92.X.X:63024 (type 00, seq 003703, ts 003200, len 000160)
Sent RTP packet to 217.92.X.X:63024 (type 00, seq 003704, ts 003360, len 000160)
Sent RTP packet to 217.92.X.X:63024 (type 00, seq 003705, ts 003520, len 000160)
- Sent RTCP SR to 217.92.X.X:63025
Our SSRC: 1381257657
Sent(NTP): 1366791258.0371900416
Sent(RTP): 3520
Sent packets: 22
Sent octets: 3520
Report block:
Fraction lost: 256
Cumulative loss: 1
IA jitter: 0.0000
Their last SR: 0
DLSR: 37978.0900 (sec)
<— SIP read from UDP:217.92.X.X:2048 —>
BYE sip:444@217.7.X.X:5060 SIP/2.0
Via: SIP/2.0/UDP 217.92.X.X:2048;branch=z9hG4bK-27jw6p9js9rg;rport
From: sip:user@217.7.X.X;tag=kf0zzs6jky
To: sip:444@217.7.X.X;tag=as1d5e4c30
Call-ID: 3c27a0e8ce1d-6kwv55572kdx
CSeq: 3 BYE
Max-Forwards: 70
Contact: sip:user@217.92.X.X:2048;reg-id=1
User-Agent: snom320/8.4.32
RTP-RxStat: Total_Rx_Pkts=22,Rx_Pkts=0,Rx_Pkts_Lost=1,Remote_Rx_Pkts_Lost=1
RTP-TxStat: Total_Tx_Pkts=427,Tx_Pkts=427,Remote_Tx_Pkts=22
Content-Length: 0
<------------->
— (12 headers 0 lines) —
Sending to 217.92.X.X:2048 (NAT)
Scheduling destruction of SIP dialog ‘3c27a0e8ce1d-6kwv55572kdx’ in 6400 ms (Method: BYE)
<— Transmitting (NAT) to 217.92.X.X:2048 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 217.92.X.X:2048;branch=z9hG4bK-27jw6p9js9rg;received=217.92.X.X;rport=2048
From: sip:user@217.7.X.X;tag=kf0zzs6jky
To: sip:444@217.7.X.X;tag=as1d5e4c30
Call-ID: 3c27a0e8ce1d-6kwv55572kdx
CSeq: 3 BYE
Server: Asterisk PBX 11.2.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0
<------------>
== Spawn extension (it-administration, 444, 3) exited non-zero on ‘SIP/user-00000007’
– Executing [h@it-administration:1] DBdeltree(“SIP/user-00000007”, “GCI/”) in new stack
– DBdeltree: family=GCI
[/code]