One-way audio when rtp ip is different from signalling ip

Hello people,

 As you will see from the trace, a call goes out from asterisk to 192.168.1.18, gets trying and answer. On the SDP its is cleary specified that the RTP IP is 192.168.1.20, but asterisk keeps sending RTPs to the signalling IP.

Is this a known bug?

Can anyone help?

fwTSCLI> rtp debug
RTP Debugging Enabled
Really destroying SIP dialog ‘HoQ05-FwG1M3f2@192.168.1.28’ Method: REGISTER
fwTS
CLI> dial 0013058883456@callshops
[Dec 23 20:55:47] WARNING[7336]: chan_oss.c:682 setformat: Unable to re-open DSP device /dev/dsp: No such file or directory
– Executing [0013058883456@callshops:1] Dial(“OSS/dsp”, “SIP/192.168.1.18/444413058883456|40”) in new stack
Audio is at 192.168.1.28 port 17320
Adding codec 0x2 (gsm) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 192.168.1.18:5060:
INVITE sip:444413058883456@192.168.1.18 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.28:5060;branch=z9hG4bK631edd89;rport
From: “asterisk” sip:asterisk@192.168.1.28;tag=as06e99d17
To: sip:444413058883456@192.168.1.18
Contact: sip:asterisk@192.168.1.28
Call-ID: 3dc51f94161036eb63407b6f1726c1b1@192.168.1.28
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Sun, 23 Dec 2007 19:55:48 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 283

v=0
o=root 7295 7295 IN IP4 192.168.1.28
s=session
c=IN IP4 192.168.1.28
t=0 0
m=audio 17320 RTP/AVP 3 0 8 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv


 -- Called 192.168.1.18/444413058883456

fwTS*CLI>
<— SIP read from 192.168.1.18:4754 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.28:5060;branch=z9hG4bK631edd89;rport=5060;received=192.168.1.28
From: “asterisk” sip:asterisk@192.168.1.28;tag=as06e99d17
To: sip:444413058883456@192.168.1.18
Call-ID: 3dc51f94161036eb63407b6f1726c1b1@192.168.1.28
CSeq: 102 INVITE
Content-Length: 0

<------------->
— (7 headers 0 lines) —
fwTSCLI>
<— SIP read from 192.168.1.18:4754 —>
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.1.28:5060;branch=z9hG4bK631edd89;rport=5060;received=192.168.1.28
From: “asterisk” sip:asterisk@192.168.1.28;tag=as06e99d17
To: sip:444413058883456@192.168.1.18;tag=00E0F51004DB21ADCC7200000A59
Call-ID: 3dc51f94161036eb63407b6f1726c1b1@192.168.1.28
CSeq: 102 INVITE
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, INFO, REFER, NOTIFY, SUBSCRIBE, UPDATE
Content-Type: application/sdp
Content-Length: 198
fwTS
CLI>
v=0
o=- 56503819500001775 1 IN IP4 192.168.1.18
s=session
c=IN IP4 192.168.1.20
t=0 0
m=audio 4970 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=ptime:20
a=sendrecv
a=rtpmap:101 telephone-event/8000

<------------->
— (9 headers 10 lines) —
Found RTP audio format 8
Found RTP audio format 101
Peer audio RTP is at port 192.168.1.20:4970
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 101
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 192.168.1.20:4970
– SIP/192.168.1.18-081f01f8 is making progress passing it to OSS/dsp
[Dec 23 20:55:48] WARNING[7411]: chan_oss.c:978 oss_indicate: Don’t know how to display condition 14 on OSS/dsp
Got RTP packet from 192.168.1.20:4970 (type 08, seq 017943, ts 1757172825, len 000160)
Got RTP packet from 192.168.1.20:4970 (type 08, seq 017944, ts 1757172985, len 000160)
Got RTP packet from 192.168.1.20:4970 (type 08, seq 017945, ts 1757173145, len 000160)
Got RTP packet from 192.168.1.20:4970 (type 08, seq 018167, ts 1757208665, len 000160)
fwTS*CLI>
<— SIP read from 192.168.1.18:4754 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.1.28:5060;branch=z9hG4bK631edd89;rport=5060;received=192.168.1.28
From: “asterisk” sip:asterisk@192.168.1.28;tag=as06e99d17
To: sip:444413058883456@192.168.1.18;tag=00E0F51004DB21ADCC7200000A59
Call-ID: 3dc51f94161036eb63407b6f1726c1b1@192.168.1.28
CSeq: 102 INVITE
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, INFO, REFER, NOTIFY, SUBSCRIBE, UPDATE
Content-Type: application/sdp
Content-Length: 198

v=0
o=- 56503819500001775 2 IN IP4 192.168.1.18
s=session
c=IN IP4 192.168.1.20
t=0 0
m=audio 4970 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=ptime:20
a=sendrecv
a=rtpmap:101 telephone-event/8000

<------------->
— (9 headers 10 lines) —
Found RTP audio format 8
Found RTP audio format 101
Peer audio RTP is at port 192.168.1.20:4970
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 101
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 192.168.1.20:4970
– SIP/192.168.1.18-081f01f8 is ringing
– SIP/192.168.1.18-081f01f8 is making progress passing it to OSS/dsp
[Dec 23 20:55:52] WARNING[7411]: chan_oss.c:978 oss_indicate: Don’t know how to display condition 14 on OSS/dsp
Got RTP packet from 192.168.1.20:4970 (type 08, seq 018168, ts 1757208825, len 000160)
Got RTP packet from 192.168.1.20:4970 (type 08, seq 018169, ts 1757208985, len 000160)
Got RTP packet from 192.168.1.20:4970 (type 08, seq 018170, ts 1757209145, len 000160)
Got RTP packet from 192.168.1.20:4970 (type 08, seq 018171, ts 1757209305, len 000160)
Got RTP packet from 192.168.1.20:4970 (type 08, seq 018172, ts 1757209465, len 000160)
Got RTP packet from 192.168.1.20:4970 (type 08, seq 018173, ts 1757209625, len 000160)
Got RTP packet from 192.168.1.20:4970 (type 08, seq 018174, ts 1757209785, len 000160)
Got RTP packet from 192.168.1.20:4970 (type 08, seq 018175, ts 1757209945, len 000160)
Got RTP packet from 192.168.1.20:4970 (type 08, seq 018176, ts 1757210105, len 000160)
Got RTP packet from 192.168.1.20:4970 (type 08, seq 018177, ts 1757210265, len 000160)
Got RTP packet from 192.168.1.20:4970 (type 08, seq 018178, ts 1757210425, len 000160)
fwTS*CLI>
<— SIP read from 192.168.1.18:4754 —>
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 192.168.1.28:5060;branch=z9hG4bK631edd89;rport=5060;received=192.168.1.28
From: “asterisk” sip:asterisk@192.168.1.28;tag=as06e99d17
To: sip:444413058883456@192.168.1.18;tag=00E0F51004DB21ADCC7200000A59
Call-ID: 3dc51f94161036eb63407b6f1726c1b1@192.168.1.28
CSeq: 102 INVITE
Contact: sip:192.168.1.18:5060
Allow-Events: refer
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, INFO, REFER, NOTIFY, SUBSCRIBE, UPDATE
Content-Type: application/sdp
Supported: 100rel, timer, replaces
Content-Length: 198

v=0
o=- 56503819500001775 3 IN IP4 192.168.1.18
s=session
c=IN IP4 192.168.1.18
t=0 0
m=audio 4970 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=ptime:20
a=sendrecv
a=rtpmap:101 telephone-event/8000

<------------->
— (12 headers 10 lines) —
Found RTP audio format 8
Found RTP audio format 101
Peer audio RTP is at port 192.168.1.18:4970
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 101
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 192.168.1.18:4970
list_route: hop: sip:192.168.1.18:5060
set_destination: Parsing sip:192.168.1.18:5060 for address/port to send to
set_destination: set destination to 192.168.1.18, port 5060
Transmitting (no NAT) to 192.168.1.18:5060:
ACK sip:192.168.1.18:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.28:5060;branch=z9hG4bK30a3155f;rport
From: “asterisk” sip:asterisk@192.168.1.28;tag=as06e99d17
To: sip:444413058883456@192.168.1.18;tag=00E0F51004DB21ADCC7200000A59
Contact: sip:asterisk@192.168.1.28
Call-ID: 3dc51f94161036eb63407b6f1726c1b1@192.168.1.28
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


-- SIP/192.168.1.18-081f01f8 answered OSS/dsp

<< Console call has been answered >>
Sent RTP DTMF packet to 192.168.1.18:4970 (type 101, seq 036865, ts 000000, len 000004)
Sent RTP DTMF packet to 192.168.1.18:4970 (type 101, seq 036866, ts 000000, len 000004)
Sent RTP DTMF packet to 192.168.1.18:4970 (type 101, seq 036867, ts 000000, len 000004)
Got RTP packet from 192.168.1.20:4970 (type 08, seq 018398, ts 1757245625, len 000160)
Sent RTP DTMF packet to 192.168.1.18:4970 (type 101, seq 036868, ts 000000, len 000004)
Got RTP packet from 192.168.1.20:4970 (type 08, seq 018399, ts 1757245785, len 000160)
Sent RTP DTMF packet to 192.168.1.18:4970 (type 101, seq 036869, ts 000000, len 000004)
Got RTP packet from 192.168.1.20:4970 (type 08, seq 018400, ts 1757245945, len 000160)
Sent RTP DTMF packet to 192.168.1.18:4970 (type 101, seq 036870, ts 000000, len 000004)
Got RTP packet from 192.168.1.20:4970 (type 08, seq 018401, ts 1757246105, len 000160)
Sent RTP DTMF packet to 192.168.1.18:4970 (type 101, seq 036871, ts 000000, len 000004)
Got RTP packet from 192.168.1.20:4970 (type 08, seq 018402, ts 1757246265, len 000160)
Sent RTP DTMF packet to 192.168.1.18:4970 (type 101, seq 036872, ts 000000, len 000004)
Got RTP packet from 192.168.1.20:4970 (type 08, seq 018403, ts 1757246425, len 000160)
Sent RTP DTMF packet to 192.168.1.18:4970 (type 101, seq 036873, ts 000000, len 000004)
Got RTP packet from 192.168.1.20:4970 (type 08, seq 018404, ts 1757246585, len 000160)
Sent RTP DTMF packet to 192.168.1.18:4970 (type 101, seq 036874, ts 000000, len 000004)
Got RTP packet from 192.168.1.20:4970 (type 08, seq 018405, ts 1757246745, len 000160)
Sent RTP DTMF packet to 192.168.1.18:4970 (type 101, seq 036875, ts 000000, len 000004)
Got RTP packet from 192.168.1.20:4970 (type 08, seq 018406, ts 1757246905, len 000160)
Sent RTP DTMF packet to 192.168.1.18:4970 (type 101, seq 036876, ts 000000, len 000004)
Got RTP packet from 192.168.1.20:4970 (type 08, seq 018407, ts 1757247065, len 000160)
Sent RTP DTMF packet to 192.168.1.18:4970 (type 101, seq 036877, ts 000000, len 000004)
Got RTP packet from 192.168.1.20:4970 (type 08, seq 018408, ts 1757247225, len 000160)
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Thanks a lot

David