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Unknown RTP codec 126 received

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Unknown RTP codec 126 received

Postby mase2hot » Sun Jan 22, 2012 4:10 am

Hi,

got an issue where I can dial out fine but there is no audio. I have udp 10001-20000 open and 5060. I dials connects but no audio either way. Search online for about 3 hours with no joy! Same trunk works fine on another system/network.

Currently disabled but on the same network / router I have asteriskNOW which works fine when ports are forwarded to it....

Thanks

Code: Select all
Found RTP audio format 0
Found RTP audio format 101
Found audio description format telephone-event for ID 101
Capabilities: us - 0x3f1fff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|ilbc|g726aal2|g722|jpeg|png|h261|h263|h263p|h264), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 192.168.1.254:7070
    -- SIP/VOIP_Unlimited-00000011 is making progress passing it to SIP/8001-00000010

<--- SIP read from 91.151.2.130:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.1.241:5060;branch=z9hG4bK18a1d85a;rport
Record-Route: <sip:91.151.2.130;lr=on;ftag=as7f884b84;did=6d7.d116efa>
Record-Route: <sip:siproxd@192.168.1.254:5060;lr>
From: "8001" <sip:01183130090@192.168.1.241>;tag=as7f884b84
To: <sip:08457203040@91.151.2.130;cpd=on>;tag=3536135231-121529
Call-ID: 70b9e200422a721b048b758866dbd0b8@192.168.1.241
CSeq: 103 INVITE
Contact: <sip:08457203040@91.151.11.20:5060>
llow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE, MESSAGE, PUBLISH
Content-Type: application/sdp
Call-Info: <sip:91.151.11.20>;method="NOTIFY;Event=telephone-event;Duration=1000"
Content-Length:   451

v=0
o=msx1-voip-unlimited-net 6853711 0 IN IP4 192.168.1.254
s=sip call
c=IN IP4 192.168.1.254
t=0 0
m=audio 7070 RTP/AVP 0 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sqn:0
a=cdsc: 1 audio RTP/AVP 0 101
a=cdsc: 3 image udptl t38
a=cpar: a=T38FaxVersion:0
a=cpar: a=T38FaxRateManagement:transferredTCF
a=cpar: a=T38FaxMaxDatagram:160
a=cpar: a=T38FaxUdpEC:t38UDPRedundancy
a=X-sqn:0
a=X-cap: 1 image udptl t38

<------------->
--- (13 headers 18 lines) ---
Found RTP audio format 0
Found RTP audio format 101
Found audio description format telephone-event for ID 101
Capabilities: us - 0x3f1fff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|ilbc|g726aal2|g722|jpeg|png|h261|h263|h263p|h264), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 192.168.1.254:7070
    -- SIP/VOIP_Unlimited-00000011 is making progress passing it to SIP/8001-00000010

<--- SIP read from 91.151.2.130:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.241:5060;branch=z9hG4bK18a1d85a;rport
Record-Route: <sip:91.151.2.130;lr=on;ftag=as7f884b84;did=6d7.d116efa>
Record-Route: <sip:siproxd@192.168.1.254:5060;lr>
From: "8001" <sip:01183130090@192.168.1.241>;tag=as7f884b84
To: <sip:08457203040@91.151.2.130;cpd=on>;tag=3536135231-121529
Call-ID: 70b9e200422a721b048b758866dbd0b8@192.168.1.241
CSeq: 103 INVITE
Contact: <sip:08457203040@91.151.11.20:5060>
Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE, MESSAGE, PUBLISH
Content-Type: application/sdp
Call-Info: <sip:91.151.11.20>;method="NOTIFY;Event=telephone-event;Duration=1000"
Content-Length:   451

v=0
o=msx1-voip-unlimited-net 6853711 0 IN IP4 192.168.1.254
s=sip call
c=IN IP4 192.168.1.254
t=0 0
m=audio 7070 RTP/AVP 0 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sqn:0
a=cdsc: 1 audio RTP/AVP 0 101
a=cdsc: 3 image udptl t38
a=cpar: a=T38FaxVersion:0
a=cpar: a=T38FaxRateManagement:transferredTCF
a=cpar: a=T38FaxMaxDatagram:160
a=cpar: a=T38FaxUdpEC:t38UDPRedundancy
a=X-sqn:0
a=X-cap: 1 image udptl t38

<------------->
--- (13 headers 18 lines) ---
Found RTP audio format 0
Found RTP audio format 101
Found audio description format telephone-event for ID 101
Capabilities: us - 0x3f1fff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|ilbc|g726aal2|g722|jpeg|png|h261|h263|h263p|h264), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 192.168.1.254:7070
list_route: hop: <sip:siproxd@192.168.1.254:5060;lr>
list_route: hop: <sip:91.151.2.130;lr=on;ftag=as7f884b84;did=6d7.d116efa>
set_destination: Parsing <sip:siproxd@192.168.1.254:5060;lr> for address/port to send to
set_destination: set destination to 192.168.1.254, port 5060
Transmitting (NAT) to 91.151.2.130:5060:
ACK sip:08457203040@91.151.11.20:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.241:5060;branch=z9hG4bK372f1b00;rport
Route: <sip:siproxd@192.168.1.254:5060;lr>,<sip:91.151.2.130;lr=on;ftag=as7f884b84;did=6d7.d116efa>
From: "8001" <sip:01183130090@192.168.1.241>;tag=as7f884b84
To: <sip:08457203040@91.151.2.130;cpd=on>;tag=3536135231-121529
Contact: <sip:01183130090@192.168.1.241>
Call-ID: 70b9e200422a721b048b758866dbd0b8@192.168.1.241
CSeq: 103 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: "8001" <sip:01183130090@192.168.1.241>;privacy=off;screen=no
Content-Length: 0


---
    -- SIP/VOIP_Unlimited-00000011 answered SIP/8001-00000010
Audio is at 192.168.1.241 port 16530
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (NAT) to 192.168.1.45:40930 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.45:40930;branch=z9hG4bK-d8754z-40a4ac4ca1c1b591-1---d8754z-;received=192.168.1.45;rport=40930
From: "Jason"<sip:8001@192.168.1.241>;tag=b83a41dd
To: <sip:908457203040@192.168.1.241>;tag=as30cf6012
Call-ID: YjgzODczNjc2YzU2OTMxMTZkNmYyZmJlNjc1ZWIwNTc.
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: <sip:908457203040@192.168.1.241>
Content-Type: application/sdp
Content-Length: 215

v=0
o=root 12338 12338 IN IP4 192.168.1.241
s=session
c=IN IP4 192.168.1.241
t=0 0
m=audio 16530 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>

<--- SIP read from 192.168.1.45:40930 --->
ACK sip:908457203040@192.168.1.241 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.45:40930;branch=z9hG4bK-d8754z-511ae220b49dc7ef-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:8001@192.168.1.45:40930>
To: <sip:908457203040@192.168.1.241>;tag=as30cf6012
From: "Jason"<sip:8001@192.168.1.241>;tag=b83a41dd
Call-ID: YjgzODczNjc2YzU2OTMxMTZkNmYyZmJlNjc1ZWIwNTc.
CSeq: 2 ACK
Proxy-Authorization: Digest username="8001",realm="asterisk",nonce="3d9e5ac2",uri="sip:908457203040@192.168.1.241",response="f100d3fe1e1b63ce6b847db3a31457dc",algorithm=MD5
User-Agent: X-Lite 4 release 4.1 stamp 63214
Content-Length: 0


<------------->
--- (11 headers 0 lines) ---
[Jan 21 11:47:12] NOTICE[30638]: rtp.c:1378 ast_rtp_read: Unknown RTP codec 126 received from '192.168.1.45'
[Jan 21 11:47:12] NOTICE[30638]: rtp.c:1378 ast_rtp_read: Unknown RTP codec 126 received from '192.168.1.45'
[Jan 21 11:47:12] NOTICE[30638]: rtp.c:1378 ast_rtp_read: Unknown RTP codec 126 received from '192.168.1.45'
Really destroying SIP dialog '0d96ea6c75ae15823414573d4651e11b@127.0.0.1' Method: REGISTER

<--- SIP read from 91.151.2.130:5060 --->
BYE sip:192.168.1.241 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.254:5060;branch=z9hG4bKc435544554cdb5843b4a62172f2d2dce
Via: SIP/2.0/UDP 91.151.2.130;branch=z9hG4bK98cf.ee0794d7.0
Via: SIP/2.0/UDP 91.151.11.20:5060;rport=5060;received=91.151.11.20;branch=z9hG4bK182fd785bd1c3eaf5f027add2e24d074
Record-Route: <sip:siproxd@192.168.1.254:5060;lr>
Record-Route: <sip:91.151.2.130;lr=on;ftag=3536135231-121529>
From: <sip:08457203040@91.151.2.130;cpd=on>;tag=3536135231-121529
To: "8001" <sip:01183130090@192.168.1.241>;tag=as7f884b84
Call-ID: 70b9e200422a721b048b758866dbd0b8@192.168.1.241
CSeq: 2 BYE
Contact: <sip:08457203040@91.151.11.20:5060>
max-forwards: 67
Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE, MESSAGE, PUBLISH
Content-Length: 0


<------------->
--- (14 headers 0 lines) ---
mase2hot
 
Posts: 4
Joined: Sun Oct 30, 2011 4:31 am
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Postby williamconley » Mon Jan 23, 2012 8:53 pm

Given the odds you are missing a sip configuration or the ports are "open" but not actually forwarded or you have a double NAT situation and do not have a router capable of fixing this problem.

Also: ports should be 10000-25000 not 10001-20000.

Did you set externip= in sip.conf?

Consider getting an external IP for the dialer if possible, post your carrier settings and:

when you post, please post your entire configuration including (but not limited to) your installation method and vicidial version with build.

this IS a requirement for posting along with reading the stickies (at the top of each forum) and the manager's manual (available on EFLO.net, both free and paid versions)

You should also post: Asterisk version, telephony hardware (model number is helpful here), cluster information if you have one, and whether any other software is installed in the box. If your installation method is "from scratch" you must post your operating system and should also post the .iso version from which you installed your original operating system.

If this is a "Cloud" or "Virtual" server, please note the technology involved along with the version of that techology (ie: VMware Server Version 2.0.2). If it is not, merely stating the Motherboard model # and CPU would be helpful.

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