Unsupported SDP media type in offer

[Jun 23 15:37:21] WARNING[17398]: chan_sip.c:5270 process_sdp: Unsupported SDP media type in offer: audio 5004 RTP/SAVP 8 3 0 18 2 4 9 97 101[/code]

I have a Grandstream 2010.
Via the webinterface I have defined the following codecs :
choice 1 pcma
choice 2 pcmu
choice 3 gsm

In sip.conf :

[grandstream]
type=friend
host=dynamic
secret=***
context=intern
canreinvite=no
qualify=yes
mailbox=10@voicemail-context
call-limit=1
accountcode=grandstream
disallow=all
allow=alaw
allow=ulaw
allow=gsm

Asterisk CLI :

[code]INVITE sip:20@192.168.1.248 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.13:5060;branch=z9hG4bK962a37dc8f9f74af
From: sip:grandstream@192.168.1.248;tag=a0435286d876c096
To: sip:20@192.168.1.248
Contact: sip:grandstream@192.168.1.13:5060;transport=udp
Supported: replaces, timer, path
Call-ID: 55ac5248810876ac@192.168.1.13
CSeq: 19256 INVITE
User-Agent: Grandstream GXP2010 1.1.6.16
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
Content-Type: application/sdp
Content-Length: 581

v=0
o=grandstream 8000 8000 IN IP4 192.168.1.13
s=SIP Call
c=IN IP4 192.168.1.13
t=0 0
m=audio 5004 RTP/SAVP 8 3 0 18 2 4 9 97 101
a=sendrecv
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:vuK0xRdYIbeSWWS8yw94rpZLOkfPIZC3mPD7zd4w
a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:AmuhiT4tOsIUg6p+1u7A0SNkLcOeZXC84xuEuqJX
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:4 G723/8000
a=rtpmap:9 G722/16000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=20
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11[/code]

Any ideas ??[/code]

One or more of:

a=rtpmap:18 G729/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:4 G723/8000 a=rtpmap:9 G722/16000 a=rtpmap:97 iLBC/8000

Best guess might be:

[quote=“david55”]One or more of:
Best guess might be:

iLBC is not supported by Asterisk, but why bother ?? There are many other codecs that can be used…

Very strange is also that I can make a phone call from my twinkle softphone, via Asterisk, to the Grandstream 2010. One can talk to another so a codec is found between the 3 SIP-clients with Asterisk in the media-path.

But when calling from the Grandstream 2010 to the twinkle softphone… I get the above error on the Asterisk CLI and on the Grandstream screen “Try other vo-coder”.

You are only demonstrating a warning, not a reason for the call to fail. I think there are some issues about Asterisk not re-opening its choice of media types when it reconfigures the RTP.

I appreciate your reply, thank you.

Indeed it is ‘just’ a warning. But the Grandstream telephone indicates on its display
"488 NOT ACCEPTABLE
Try other vocoder"

So the reason why the call isn’t established, I think, is because of a codec-issue…

Can you advise ?

Need the sip set debug trace up to the point where the 488 response is generated/received.

<— SIP read from 192.168.1.13:5060 —>
INVITE sip:20@192.168.1.248 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.13:5060;branch=z9hG4bK2c53432a04169c8f
From: sip:grandstream@192.168.1.248;tag=00a56a43a5305e76
To: sip:20@192.168.1.248
Contact: sip:grandstream@192.168.1.13:5060;transport=udp
Supported: replaces, timer, path
Call-ID: b3be90bbee30f307@192.168.1.13
CSeq: 13102 INVITE
User-Agent: Grandstream GXP2010 1.1.6.16
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
Content-Type: application/sdp
Content-Length: 581

v=0
o=grandstream 8000 8000 IN IP4 192.168.1.13
s=SIP Call
c=IN IP4 192.168.1.13
t=0 0
m=audio 5004 RTP/SAVP 8 0 3 18 2 4 9 97 101
a=sendrecv
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:MG3Wnd/3zNcMrORVbPySyhxVkwgXo4Yv9jap+eAW
a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:4mSvAyxY1e6WeSLpNo/s0wkfBAkYuPBheBSq1ihh
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:18 G729/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:4 G723/8000
a=rtpmap:9 G722/16000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=20
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11

<------------->
— (13 headers 21 lines) —
Sending to 192.168.1.13 : 5060 (no NAT)
Using INVITE request as basis request - b3be90bbee30f307@192.168.1.13

<— Reliably Transmitting (no NAT) to 192.168.1.13:5060 —>
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.1.13:5060;branch=z9hG4bK2c53432a04169c8f;received=192.168.1.13
From: sip:grandstream@192.168.1.248;tag=00a56a43a5305e76
To: sip:20@192.168.1.248;tag=as1afd302f
Call-ID: b3be90bbee30f307@192.168.1.13
CSeq: 13102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm=“asterisk-jocan”, nonce="555d7311"
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘b3be90bbee30f307@192.168.1.13’ in 32000 ms (Method: INVITE)
Found user 'grandstream’
asterisk*CLI>
<— SIP read from 192.168.1.13:5060 —>
ACK sip:20@192.168.1.248 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.13:5060;branch=z9hG4bK2c53432a04169c8f
From: sip:grandstream@192.168.1.248;tag=00a56a43a5305e76
To: sip:20@192.168.1.248;tag=as1afd302f
Contact: sip:grandstream@192.168.1.13:5060;transport=udp
Supported: path
Call-ID: b3be90bbee30f307@192.168.1.13
CSeq: 13102 ACK
User-Agent: Grandstream GXP2010 1.1.6.16
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
Content-Length: 0

<------------->
— (12 headers 0 lines) —
asterisk*CLI>
<— SIP read from 192.168.1.13:5060 —>
INVITE sip:20@192.168.1.248 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.13:5060;branch=z9hG4bKf2de3e3a9b3abcec
From: sip:grandstream@192.168.1.248;tag=00a56a43a5305e76
To: sip:20@192.168.1.248
Contact: sip:grandstream@192.168.1.13:5060;transport=udp
Supported: replaces, timer, path
Proxy-Authorization: Digest username=“grandstream”, realm=“asterisk-jocan”, algorithm=MD5, uri="sip:20@192.168.1.248", nonce=“555d7311”, response="c5f14dcfcc15859f0d55335fae76a316"
Call-ID: b3be90bbee30f307@192.168.1.13
CSeq: 13103 INVITE
User-Agent: Grandstream GXP2010 1.1.6.16
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
Content-Type: application/sdp
Content-Length: 581

v=0
o=grandstream 8000 8001 IN IP4 192.168.1.13
s=SIP Call
c=IN IP4 192.168.1.13
t=0 0
m=audio 5004 RTP/SAVP 8 0 3 18 2 4 9 97 101
a=sendrecv
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:MG3Wnd/3zNcMrORVbPySyhxVkwgXo4Yv9jap+eAW
a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:4mSvAyxY1e6WeSLpNo/s0wkfBAkYuPBheBSq1ihh
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:18 G729/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:4 G723/8000
a=rtpmap:9 G722/16000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=20
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11

<------------->
— (14 headers 21 lines) —
Sending to 192.168.1.13 : 5060 (no NAT)
Using INVITE request as basis request - b3be90bbee30f307@192.168.1.13
Found user ‘grandstream’
[Jun 23 16:42:40] WARNING[17504]: chan_sip.c:5270 process_sdp: Unsupported SDP media type in offer: audio 5004 RTP/SAVP 8 0 3 18 2 4 9 97 101

<— Reliably Transmitting (no NAT) to 192.168.1.13:5060 —>
SIP/2.0 488 Not acceptable here
Via: SIP/2.0/UDP 192.168.1.13:5060;branch=z9hG4bKf2de3e3a9b3abcec;received=192.168.1.13
From: sip:grandstream@192.168.1.248;tag=00a56a43a5305e76
To: sip:20@192.168.1.248;tag=as1afd302f
Call-ID: b3be90bbee30f307@192.168.1.13
CSeq: 13103 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0

You need to turn up the verbosity.

asterisk*CLI> core set verbose 99 Verbosity was 25 and is now 99

Here we go :

Connected to Asterisk 1.4.25.1 currently running on asterisk (pid = 12780)
Verbosity is at least 99
Core debug is at least 25
asterisk*CLI>
<— SIP read from 192.168.1.13:5060 —>
INVITE sip:20@192.168.1.248 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.13:5060;branch=z9hG4bK52feb2fe04930b5a
From: sip:grandstream@192.168.1.248;tag=71360aaf46d50083
To: sip:20@192.168.1.248
Contact: sip:grandstream@192.168.1.13:5060;transport=udp
Supported: replaces, timer, path
Call-ID: 49c8921ff0eebfbb@192.168.1.13
CSeq: 50555 INVITE
User-Agent: Grandstream GXP2010 1.1.6.16
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
Content-Type: application/sdp
Content-Length: 581

v=0
o=grandstream 8000 8000 IN IP4 192.168.1.13
s=SIP Call
c=IN IP4 192.168.1.13
t=0 0
m=audio 5004 RTP/SAVP 8 0 3 18 2 4 9 97 101
a=sendrecv
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:ckes5wzRCfuIiZAGQM/8qCaQ3P8Xip0p8I96zlXD
a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:sNVe0OMtQAp9kBzjgm3OO7aXtbWUlZwSTNsIwD6c
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:18 G729/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:4 G723/8000
a=rtpmap:9 G722/16000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=20
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11

<------------->
— (13 headers 21 lines) —
Sending to 192.168.1.13 : 5060 (no NAT)
Using INVITE request as basis request - 49c8921ff0eebfbb@192.168.1.13

<— Reliably Transmitting (no NAT) to 192.168.1.13:5060 —>
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.1.13:5060;branch=z9hG4bK52feb2fe04930b5a;received=192.168.1.13
From: sip:grandstream@192.168.1.248;tag=71360aaf46d50083
To: sip:20@192.168.1.248;tag=as0d098212
Call-ID: 49c8921ff0eebfbb@192.168.1.13
CSeq: 50555 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm=“asterisk-jocan”, nonce="74a7ea42"
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘49c8921ff0eebfbb@192.168.1.13’ in 32000 ms (Method: INVITE)
Found user 'grandstream’
asterisk*CLI>
<— SIP read from 192.168.1.13:5060 —>
ACK sip:20@192.168.1.248 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.13:5060;branch=z9hG4bK52feb2fe04930b5a
From: sip:grandstream@192.168.1.248;tag=71360aaf46d50083
To: sip:20@192.168.1.248;tag=as0d098212
Contact: sip:grandstream@192.168.1.13:5060;transport=udp
Supported: path
Call-ID: 49c8921ff0eebfbb@192.168.1.13
CSeq: 50555 ACK
User-Agent: Grandstream GXP2010 1.1.6.16
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
Content-Length: 0

<------------->
— (12 headers 0 lines) —
asterisk*CLI>
<— SIP read from 192.168.1.13:5060 —>
INVITE sip:20@192.168.1.248 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.13:5060;branch=z9hG4bKb918a0ee9bde174a
From: sip:grandstream@192.168.1.248;tag=71360aaf46d50083
To: sip:20@192.168.1.248
Contact: sip:grandstream@192.168.1.13:5060;transport=udp
Supported: replaces, timer, path
Proxy-Authorization: Digest username=“grandstream”, realm=“asterisk-jocan”, algorithm=MD5, uri="sip:20@192.168.1.248", nonce=“74a7ea42”, response="73c382dfef046527bb6b26b888d3f1ca"
Call-ID: 49c8921ff0eebfbb@192.168.1.13
CSeq: 50556 INVITE
User-Agent: Grandstream GXP2010 1.1.6.16
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
Content-Type: application/sdp
Content-Length: 581

v=0
o=grandstream 8000 8001 IN IP4 192.168.1.13
s=SIP Call
c=IN IP4 192.168.1.13
t=0 0
m=audio 5004 RTP/SAVP 8 0 3 18 2 4 9 97 101
a=sendrecv
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:ckes5wzRCfuIiZAGQM/8qCaQ3P8Xip0p8I96zlXD
a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:sNVe0OMtQAp9kBzjgm3OO7aXtbWUlZwSTNsIwD6c
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:18 G729/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:4 G723/8000
a=rtpmap:9 G722/16000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=20
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11

<------------->
— (14 headers 21 lines) —
Sending to 192.168.1.13 : 5060 (no NAT)
Using INVITE request as basis request - 49c8921ff0eebfbb@192.168.1.13
Found user ‘grandstream’
[Jun 23 16:58:42] WARNING[17504]: chan_sip.c:5270 process_sdp: Unsupported SDP media type in offer: audio 5004 RTP/SAVP 8 0 3 18 2 4 9 97 101

<— Reliably Transmitting (no NAT) to 192.168.1.13:5060 —>
SIP/2.0 488 Not acceptable here
Via: SIP/2.0/UDP 192.168.1.13:5060;branch=z9hG4bKb918a0ee9bde174a;received=192.168.1.13
From: sip:grandstream@192.168.1.248;tag=71360aaf46d50083
To: sip:20@192.168.1.248;tag=as0d098212
Call-ID: 49c8921ff0eebfbb@192.168.1.13
CSeq: 50556 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘49c8921ff0eebfbb@192.168.1.13’ in 32000 ms (Method: INVITE)
asterisk*CLI>
<— SIP read from 192.168.1.13:5060 —>
ACK sip:20@192.168.1.248 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.13:5060;branch=z9hG4bKb918a0ee9bde174a
From: sip:grandstream@192.168.1.248;tag=71360aaf46d50083
To: sip:20@192.168.1.248;tag=as0d098212
Contact: sip:grandstream@192.168.1.13:5060;transport=udp
Supported: path
Proxy-Authorization: Digest username=“grandstream”, realm=“asterisk-jocan”, algorithm=MD5, uri="sip:20@192.168.1.248", nonce=“74a7ea42”, response="73c382dfef046527bb6b26b888d3f1ca"
Call-ID: 49c8921ff0eebfbb@192.168.1.13
CSeq: 50556 ACK
User-Agent: Grandstream GXP2010 1.1.6.16
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
Content-Length: 0

<------------->
— (13 headers 0 lines) —
asterisk*CLI> exit

When I do this, I get some information on how it is interpreting the messages. Maybe it isn’t verbose that controls this, but its something that I always do, so I don’t need to think about it.

I think some of the relevant messages are at NOTICE level.

In logger.conf I have :

console => notice,warning,error

So that’s all the notice you can get, I guess…

Actually, most of the interesting stuff is at debug level 3.

asteriskCLI> core set debug 3
Core debug was 5 and is now 3
asterisk
CLI>
<— SIP read from 192.168.1.13:5060 —>
INVITE sip:20@192.168.1.248 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.13:5060;branch=z9hG4bKadc677f67e732d08
From: sip:grandstream@192.168.1.248;tag=74de4ece283a883a
To: sip:20@192.168.1.248
Contact: sip:grandstream@192.168.1.13:5060;transport=udp
Supported: replaces, timer, path
Call-ID: 8c1594e7e6e7ef9b@192.168.1.13
CSeq: 40411 INVITE
User-Agent: Grandstream GXP2010 1.1.6.16
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
Content-Type: application/sdp
Content-Length: 581

v=0
o=grandstream 8000 8000 IN IP4 192.168.1.13
s=SIP Call
c=IN IP4 192.168.1.13
t=0 0
m=audio 5004 RTP/SAVP 8 0 3 18 2 4 9 97 101
a=sendrecv
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:0t1biyJd7Ch5uaSuERaMOfj3dPOij5ltDhcUmMBT
a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:6VUcz8wuW5WCjGABCgeLs0aXYPyFxbC6HSKYRQQD
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:18 G729/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:4 G723/8000
a=rtpmap:9 G722/16000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=20
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11

<------------->
— (13 headers 21 lines) —
Sending to 192.168.1.13 : 5060 (no NAT)
Using INVITE request as basis request - 8c1594e7e6e7ef9b@192.168.1.13

<— Reliably Transmitting (no NAT) to 192.168.1.13:5060 —>
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.1.13:5060;branch=z9hG4bKadc677f67e732d08;received=192.168.1.13
From: sip:grandstream@192.168.1.248;tag=74de4ece283a883a
To: sip:20@192.168.1.248;tag=as544f6459
Call-ID: 8c1594e7e6e7ef9b@192.168.1.13
CSeq: 40411 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm=“asterisk-jocan”, nonce="393c64b3"
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘8c1594e7e6e7ef9b@192.168.1.13’ in 32000 ms (Method: INVITE)
Found user 'grandstream’
asterisk*CLI>
<— SIP read from 192.168.1.13:5060 —>
ACK sip:20@192.168.1.248 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.13:5060;branch=z9hG4bKadc677f67e732d08
From: sip:grandstream@192.168.1.248;tag=74de4ece283a883a
To: sip:20@192.168.1.248;tag=as544f6459
Contact: sip:grandstream@192.168.1.13:5060;transport=udp
Supported: path
Call-ID: 8c1594e7e6e7ef9b@192.168.1.13
CSeq: 40411 ACK
User-Agent: Grandstream GXP2010 1.1.6.16
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
Content-Length: 0

<------------->
— (12 headers 0 lines) —
asterisk*CLI>
<— SIP read from 192.168.1.13:5060 —>
INVITE sip:20@192.168.1.248 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.13:5060;branch=z9hG4bKad969576c5c63263
From: sip:grandstream@192.168.1.248;tag=74de4ece283a883a
To: sip:20@192.168.1.248
Contact: sip:grandstream@192.168.1.13:5060;transport=udp
Supported: replaces, timer, path
Proxy-Authorization: Digest username=“grandstream”, realm=“asterisk-jocan”, algorithm=MD5, uri="sip:20@192.168.1.248", nonce=“393c64b3”, response="95751fe693dbdb680eedc73171db821c"
Call-ID: 8c1594e7e6e7ef9b@192.168.1.13
CSeq: 40412 INVITE
User-Agent: Grandstream GXP2010 1.1.6.16
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
Content-Type: application/sdp
Content-Length: 581

v=0
o=grandstream 8000 8001 IN IP4 192.168.1.13
s=SIP Call
c=IN IP4 192.168.1.13
t=0 0
m=audio 5004 RTP/SAVP 8 0 3 18 2 4 9 97 101
a=sendrecv
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:0t1biyJd7Ch5uaSuERaMOfj3dPOij5ltDhcUmMBT
a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:6VUcz8wuW5WCjGABCgeLs0aXYPyFxbC6HSKYRQQD
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:18 G729/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:4 G723/8000
a=rtpmap:9 G722/16000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=20
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11

<------------->
— (14 headers 21 lines) —
Sending to 192.168.1.13 : 5060 (no NAT)
Using INVITE request as basis request - 8c1594e7e6e7ef9b@192.168.1.13
Found user ‘grandstream’
[Jun 23 17:40:52] WARNING[17504]: chan_sip.c:5270 process_sdp: Unsupported SDP media type in offer: audio 5004 RTP/SAVP 8 0 3 18 2 4 9 97 101

<— Reliably Transmitting (no NAT) to 192.168.1.13:5060 —>
SIP/2.0 488 Not acceptable here
Via: SIP/2.0/UDP 192.168.1.13:5060;branch=z9hG4bKad969576c5c63263;received=192.168.1.13
From: sip:grandstream@192.168.1.248;tag=74de4ece283a883a
To: sip:20@192.168.1.248;tag=as544f6459
Call-ID: 8c1594e7e6e7ef9b@192.168.1.13
CSeq: 40412 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘8c1594e7e6e7ef9b@192.168.1.13’ in 32000 ms (Method: INVITE)
asterisk*CLI>
<— SIP read from 192.168.1.13:5060 —>
ACK sip:20@192.168.1.248 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.13:5060;branch=z9hG4bKad969576c5c63263
From: sip:grandstream@192.168.1.248;tag=74de4ece283a883a
To: sip:20@192.168.1.248;tag=as544f6459
Contact: sip:grandstream@192.168.1.13:5060;transport=udp
Supported: path
Proxy-Authorization: Digest username=“grandstream”, realm=“asterisk-jocan”, algorithm=MD5, uri="sip:20@192.168.1.248", nonce=“393c64b3”, response="95751fe693dbdb680eedc73171db821c"
Call-ID: 8c1594e7e6e7ef9b@192.168.1.13
CSeq: 40412 ACK
User-Agent: Grandstream GXP2010 1.1.6.16
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
Content-Length: 0

You need to enable debug to whatever log you are cutting and pasting. Basically you are trying to trigger this code:

ast_debug(3, "*** Our native formats are %s \n", ast_getformatname_multiple(buf, SIPBUFSIZE, tmp->nativeformats)); ast_debug(3, "*** Joint capabilities are %s \n", ast_getformatname_multiple(buf, SIPBUFSIZE, i->jointcapability)); ast_debug(3, "*** Our capabilities are %s \n", ast_getformatname_multiple(buf, SIPBUFSIZE, i->capability));

I am cutting and pasting from the Asterisk CLI.

Your code, do I need to paste this into my dialplan ?

No. The code is part of chan_sip.c and is what writes the debugging trace information that will tell you what Asterisk thinks is happening with the codecs. Until you can get a trace that includes what that code writes, there is little point in posting it to the forum.

For information, this is what I have in logger.conf:

;debug => debug console => notice,warning,error ;console => notice,warning,error,debug messages => notice,warning,error full => notice,warning,error,debug,verbose

I’d normally use /var/log/asterisk/full for bug reports.

I probably normally set debugging to level 5.

I have :

debug => debug console => notice,warning,error ;console => notice,warning,error,debug messages => notice,warning,error verbose => verbose ;full => notice,warning,error,debug,verbose

I have :

if (option_debug > 2) { char buf[SIPBUFSIZE]; ast_log(LOG_DEBUG, "*** Our native formats are %s \n", ast_getformatname_multiple(buf, SIPBUFSIZE, tmp->nativeformats)); ast_log(LOG_DEBUG, "*** Joint capabilities are %s \n", ast_getformatname_multiple(buf, SIPBUFSIZE, i->jointcapability)); ast_log(LOG_DEBUG, "*** Our capabilities are %s \n", ast_getformatname_multiple(buf, SIPBUFSIZE, i->capability)); ast_log(LOG_DEBUG, "*** AST_CODEC_CHOOSE formats are %s \n", ast_getformatname_multiple(buf, SIPBUFSIZE, ast_codec_choose(&i->prefs, what, 1 ))); if (i->prefcodec) ast_log(LOG_DEBUG, "*** Our preferred formats from the incoming channel are %s \n", ast_getformatname_multiple(buf, SIPBUFSIZE, i->p refcodec)); }

You should look at /var/log/asterisk/debug

The exact code probably varies from version to version; it is the output of that code which ought to tell us what Asterisk thinks is happening.

On CLI :

Connected to Asterisk 1.4.25.1 currently running on asterisk (pid = 17936) Verbosity is at least 25 Core debug is at least 5 [Jun 23 19:29:05] WARNING[17971]: chan_sip.c:5270 process_sdp: Unsupported SDP media type in offer: audio 5004 RTP/SAVP 8 0 3 18 2 4 9 97 101 asterisk*CLI> exit

Log debug :

[Jun 23 19:29:05] DEBUG[17971] chan_sip.c: = No match Their Call ID: 20bca78b08222d356cd89a777a11be24@127.0.0.1 Their Tag 470f6df9d438ef0e611aed43a3c90fcf.a
f5f Our tag: as44e96dfb
[Jun 23 19:29:05] DEBUG[17971] chan_sip.c: Setting NAT on RTP to Off
[Jun 23 19:29:05] DEBUG[17971] chan_sip.c: Allocating new SIP dialog for 3caeb2b227d6096b@192.168.1.13 - INVITE (With RTP)
[Jun 23 19:29:05] DEBUG[17971] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE
[Jun 23 19:29:05] DEBUG[17971] chan_sip.c: Setting NAT on RTP to Off
[Jun 23 19:29:05] DEBUG[17971] chan_sip.c: = Found Their Call ID: 3caeb2b227d6096b@192.168.1.13 Their Tag 80dc1a9fcd9f4353 Our tag: as21a20e17
[Jun 23 19:29:05] DEBUG[17971] chan_sip.c: **** Received ACK (6) - Command in SIP ACK
[Jun 23 19:29:05] DEBUG[17971] chan_sip.c: Stopping retransmission on ‘3caeb2b227d6096b@192.168.1.13’ of Response 17053: Match Found
[Jun 23 19:29:05] DEBUG[17971] chan_sip.c: = Found Their Call ID: 3caeb2b227d6096b@192.168.1.13 Their Tag 80dc1a9fcd9f4353 Our tag: as21a20e17
[Jun 23 19:29:05] DEBUG[17971] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE
[Jun 23 19:29:05] DEBUG[17971] chan_sip.c: Setting NAT on RTP to Off
[Jun 23 19:29:05] DEBUG[17971] chan_sip.c: No compatible codecs for this SIP call.
[Jun 23 19:29:05] DEBUG[17971] chan_sip.c: SIP message could not be handled, bad request: 3caeb2b227d6096b@192.168.1.13

[Jun 23 19:29:05] DEBUG[17971] chan_sip.c: = Found Their Call ID: 3caeb2b227d6096b@192.168.1.13 Their Tag 80dc1a9fcd9f4353 Our tag: as21a20e17
[Jun 23 19:29:05] DEBUG[17971] chan_sip.c: **** Received ACK (6) - Command in SIP ACK
[Jun 23 19:29:05] DEBUG[17971] chan_sip.c: Stopping retransmission on ‘3caeb2b227d6096b@192.168.1.13’ of Response 17054: Match Found

Solution @ issues.asterisk.org/view.php?id=15384

! RESOLVED !

Thanks for the help, david55.