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Unsupported media type 415 . internal calls


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Hi,

In my pbx I tried changing the codec priority in system settings, allowing only g711u and g711a. When I did a test call from an extension (who has the same codec enabled and many more) I receive error 415 Unsupported media type.

 

I then removed the changes I made, but the error remains. I tried to reboot the pbx with no luck

 

Any help?

 

I attach here the log from a test call

 

INVITE sip:8877@176.50.240.2 SIP/2.0
Via: SIP/2.0/UDP 85.50.240.22:53137;branch=z9hG4bK80b87a1db7d0e1118ab1001e9030b32e;rport
From: "43402" <sip:43402@176.50.240.2>;tag=3448809797
To: <sip:8877@176.50.240.2>
Call-ID: 80B87A1D-B7D0-E111-8AB0-001E9030B32E@85.50.240.22
CSeq: 8 INVITE
Contact: <sip:43402@85.50.240.22:53137>
Content-Type: application/sdp
Allow: INVITE, OPTIONS, ACK, BYE, CANCEL, INFO, NOTIFY, MESSAGE, UPDATE
Max-Forwards: 70
Supported: 100rel, replaces, from-change
User-Agent: SIPPER for PhonerLite
P-Preferred-Identity: <sip:43402@176.50.240.2>
Content-Length: 394

v=0
o=- 1452070351 0 IN IP4 192.168.1.5
s=SIPPER for PhonerLite
c=IN IP4 192.168.1.5
t=0 0
m=audio 53139 RTP/AVP 0 8 2 3 97 110 111 9 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:3 GSM/8000
a=rtpmap:97 iLBC/8000
a=rtpmap:110 speex/8000
a=rtpmap:111 speex/16000
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv
[5] 2012/07/20 11:00:24:	SIP Tx udp:85.50.240.22:53137:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 85.50.240.22:53137;branch=z9hG4bK80b87a1db7d0e1118ab1001e9030b32e;rport=53137
From: "43402" <sip:43402@176.50.240.2>;tag=3448809797
To: <sip:8877@176.50.240.2>;tag=c0c8306a6c
Call-ID: 80B87A1D-B7D0-E111-8AB0-001E9030B32E@85.50.240.22
CSeq: 8 INVITE
Content-Length: 0

[5] 2012/07/20 11:00:24:	SIP Tx udp:85.50.240.22:53137:
SIP/2.0 401 Authentication Required
Via: SIP/2.0/UDP 85.50.240.22:53137;branch=z9hG4bK80b87a1db7d0e1118ab1001e9030b32e;rport=53137
From: "43402" <sip:43402@176.50.240.2>;tag=3448809797
To: <sip:8877@176.50.240.2>;tag=c0c8306a6c
Call-ID: 80B87A1D-B7D0-E111-8AB0-001E9030B32E@85.50.240.22
CSeq: 8 INVITE
User-Agent: snomONE/4.5.0.1075 Delta Aurigids
WWW-Authenticate: Digest realm="176.50.240.2",nonce="6ae674e91ce208ae882342ff67954e2a",domain="sip:8877@176.50.240.2",algorithm=MD5
Content-Length: 0

[5] 2012/07/20 11:00:24:	SIP Rx udp:85.50.240.22:53137:
ACK sip:8877@176.50.240.2 SIP/2.0
Via: SIP/2.0/UDP 85.50.240.22:53137;branch=z9hG4bK80b87a1db7d0e1118ab1001e9030b32e;rport
From: "43402" <sip:43402@176.50.240.2>;tag=3448809797
To: <sip:8877@176.50.240.2>;tag=c0c8306a6c
Call-ID: 80B87A1D-B7D0-E111-8AB0-001E9030B32E@85.50.240.22
CSeq: 8 ACK
Max-Forwards: 70
Content-Length: 0

[5] 2012/07/20 11:00:24:	SIP Rx udp:85.50.240.22:53137:
INVITE sip:8877@176.50.240.2 SIP/2.0
Via: SIP/2.0/UDP 85.50.240.22:53137;branch=z9hG4bK803f0e21b7d0e1118ab1001e9030b32e;rport
From: "43402" <sip:43402@176.50.240.2>;tag=3448809797
To: <sip:8877@176.50.240.2>
Call-ID: 80B87A1D-B7D0-E111-8AB0-001E9030B32E@85.50.240.22
CSeq: 9 INVITE
Contact: <sip:43402@85.50.240.22:53137>
Authorization: Digest username="43402", realm="176.50.240.2", nonce="6ae674e91ce208ae882342ff67954e2a", uri="sip:8877@176.50.240.2", response="d47ab92ac330ba427d2b0770a3f555a1", algorithm=MD5
Content-Type: application/sdp
Allow: INVITE, OPTIONS, ACK, BYE, CANCEL, INFO, NOTIFY, MESSAGE, UPDATE
Max-Forwards: 70
Supported: 100rel, replaces, from-change
User-Agent: SIPPER for PhonerLite
P-Preferred-Identity: <sip:43402@176.50.240.2>
Content-Length: 394

v=0
o=- 1452070351 0 IN IP4 192.168.1.5
s=SIPPER for PhonerLite
c=IN IP4 192.168.1.5
t=0 0
m=audio 53139 RTP/AVP 0 8 2 3 97 110 111 9 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:3 GSM/8000
a=rtpmap:97 iLBC/8000
a=rtpmap:110 speex/8000
a=rtpmap:111 speex/16000
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv
[5] 2012/07/20 11:00:24:	SIP Tx udp:85.50.240.22:53137:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 85.50.240.22:53137;branch=z9hG4bK803f0e21b7d0e1118ab1001e9030b32e;rport=53137
From: "43402" <sip:43402@176.50.240.2>;tag=3448809797
To: <sip:8877@176.50.240.2>;tag=c0c8306a6c
Call-ID: 80B87A1D-B7D0-E111-8AB0-001E9030B32E@85.50.240.22
CSeq: 9 INVITE
Content-Length: 0

[5] 2012/07/20 11:00:24:	SIP Tx udp:85.50.240.22:53137:
SIP/2.0 415 Unsupported Media Type
Via: SIP/2.0/UDP 85.50.240.22:53137;branch=z9hG4bK803f0e21b7d0e1118ab1001e9030b32e;rport=53137
From: "43402" <sip:43402@176.50.240.2>;tag=3448809797
To: <sip:8877@176.50.240.2>;tag=c0c8306a6c
Call-ID: 80B87A1D-B7D0-E111-8AB0-001E9030B32E@85.50.240.22
CSeq: 9 INVITE
Contact: <sip:43402@176.50.240.2:5060>
Supported: 100rel, replaces, norefersub
Allow-Events: refer
Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE
Accept: application/sdp
User-Agent: snomONE/4.5.0.1075 Delta Aurigids
Content-Length: 0

[5] 2012/07/20 11:00:24:	SIP Rx udp:85.50.240.22:53137:
ACK sip:8877@176.50.240.2 SIP/2.0
Via: SIP/2.0/UDP 85.50.240.22:53137;branch=z9hG4bK803f0e21b7d0e1118ab1001e9030b32e;rport
From: "43402" <sip:43402@176.50.240.2>;tag=3448809797
To: <sip:8877@176.50.240.2>;tag=c0c8306a6c
Call-ID: 80B87A1D-B7D0-E111-8AB0-001E9030B32E@85.50.240.22
CSeq: 9 ACK
Authorization: Digest username="43402", realm="176.50.240.2", nonce="6ae674e91ce208ae882342ff67954e2a", uri="sip:8877@176.50.240.2", response="d47ab92ac330ba427d2b0770a3f555a1", algorithm=MD5
Max-Forwards: 70
Content-Length: 0

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Check the "SIPPER for PhonerLite" codec and see if it's using a different one then what is supported on the pbx..

 

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:2 G726-32/8000

a=rtpmap:3 GSM/8000

a=rtpmap:97 iLBC/8000

a=rtpmap:110 speex/8000

a=rtpmap:111 speex/16000

a=rtpmap:9 G722/8000

 

these are the one configured

 

SmonOne has all the default codec enabled, which should comprehend g711u,g711a

 

Why do I receive the error?

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Hmm. Do you also see the traffic on the other side of the B2BUA? Maybe the codec negotiation runs into a problem on the other leg, and the PBX just relay the message "415 Unsupported Media Type". It can also happen if the PBX proposes SRTP, but does not get the SRTP key back.

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Hmm. Do you also see the traffic on the other side of the B2BUA? Maybe the codec negotiation runs into a problem on the other leg, and the PBX just relay the message "415 Unsupported Media Type". It can also happen if the PBX proposes SRTP, but does not get the SRTP key back.

 

Fixed the issue, but don't know the cause

 

how I fixed:

 

removed supported codec from Domain --> Settings --> Port

Save

Reboot pbx

insert supported codec in Domain --> settings --> Port

Save

Reboot pbx

 

I honestly don't know if all the steps were required. But at least I have the pbx working again.

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