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Din Assegaf

2012-Jan-28 16:22 UTC

[asterisk-users] process_sdp: Unsupported SDP media type in offer: audio , Failing due to no acceptable offer found

Hi All,

I'm trying to upgrade asterisk server to 1.8.x from my asterisk 1.6,

But when making A Call from SIP Client, I got cli Warning ... and no call
has been made.

My Sip Client is using lib java peers client http://peers.sourceforge.net/
with standard codec PCMU/PCMA

[Jan 28 23:03:32] WARNING[1654]: chan_sip.c:8942 process_sdp: Unsupported
SDP media type in offer: audio 0 RTP/AVP 0 8 101
[Jan 28 23:03:32] WARNING[1654]: chan_sip.c:9029 process_sdp: Failing due
to no acceptable offer found

the strange thing is when using asterisk 1.6, is normal,
when using asterisk 1.8.x and using another client like Ekiga is normal too,

so the main problem is when using combination Asterisk 1.8.x (I have tried
the last 1.8.9.0 also), and using lib java peers client, it is fail.


This is the Full DEBUG log.

I dont know what else to do, googling found no one with the similar problem.


[Jan 28 23:03:32] DEBUG[1654]: chan_sip.c:8014 find_call: = Looking for
 Call ID: hDVA1Kyx-1327766611250 at lucidesktop.lan (Checking From) --From tag
grUqFtoE --To-tag
[Jan 28 23:03:32] DEBUG[1654]: acl.c:728 ast_ouraddrfor: For destination
'192.168.2.159', our source address is '192.168.2.172'.
[Jan 28 23:03:32] DEBUG[1654]: chan_sip.c:3482 ast_sip_ouraddrfor: Setting
SIP_TRANSPORT_UDP with address 192.168.2.172:5060
[Jan 28 23:03:32] DEBUG[1654]: chan_sip.c:7694 sip_alloc: Allocating new
SIP dialog for hDVA1Kyx-1327766611250 at lucidesktop.lan - INVITE (No RTP)
[Jan 28 23:03:32] DEBUG[1654]: chan_sip.c:24907 handle_incoming: ****
Received INVITE (5) - Command in SIP INVITE
[Jan 28 23:03:32] DEBUG[1654]: netsock2.c:134 ast_sockaddr_split_hostport:
Splitting '192.168.2.159:5062' into...
[Jan 28 23:03:32] DEBUG[1654]: netsock2.c:188 ast_sockaddr_split_hostport:
...host '192.168.2.159' and port '5062'.
[Jan 28 23:03:32] DEBUG[1654]: netsock2.c:134 ast_sockaddr_split_hostport:
Splitting '192.168.2.172' into...
[Jan 28 23:03:32] DEBUG[1654]: netsock2.c:188 ast_sockaddr_split_hostport:
...host '192.168.2.172' and port ''.
[Jan 28 23:03:32] DEBUG[1654]: chan_sip.c:3328 __sip_xmit: Trying to put
'SIP/2.0 401' onto UDP socket destined for 192.168.2.159:5062
[Jan 28 23:03:32] DEBUG[1654]: chan_sip.c:8014 find_call: = Looking for
 Call ID: hDVA1Kyx-1327766611250 at lucidesktop.lan (Checking From) --From tag
grUqFtoE --To-tag as5d163902
[Jan 28 23:03:32] DEBUG[1654]: chan_sip.c:24907 handle_incoming: ****
Received ACK (6) - Command in SIP ACK
[Jan 28 23:03:32] DEBUG[1654]: chan_sip.c:4013 __sip_ack: Stopping
retransmission on 'hDVA1Kyx-1327766611250 at lucidesktop.lan' of
Response 5:
Match Found
[Jan 28 23:03:32] DEBUG[1654]: chan_sip.c:8014 find_call: = Looking for
 Call ID: hDVA1Kyx-1327766611250 at lucidesktop.lan (Checking From) --From tag
grUqFtoE --To-tag
[Jan 28 23:03:32] DEBUG[1654]: netsock2.c:134 ast_sockaddr_split_hostport:
Splitting '192.168.2.172' into...
[Jan 28 23:03:32] DEBUG[1654]: netsock2.c:188 ast_sockaddr_split_hostport:
...host '192.168.2.172' and port ''.
[Jan 28 23:03:32] DEBUG[1654]: netsock2.c:134 ast_sockaddr_split_hostport:
Splitting '192.168.2.172' into...
[Jan 28 23:03:32] DEBUG[1654]: netsock2.c:188 ast_sockaddr_split_hostport:
...host '192.168.2.172' and port ''.
[Jan 28 23:03:32] DEBUG[1654]: chan_sip.c:24907 handle_incoming: ****
Received INVITE (5) - Command in SIP INVITE
[Jan 28 23:03:32] DEBUG[1654]: netsock2.c:134 ast_sockaddr_split_hostport:
Splitting '192.168.2.159:5062' into...
[Jan 28 23:03:32] DEBUG[1654]: netsock2.c:188 ast_sockaddr_split_hostport:
...host '192.168.2.159' and port '5062'.
[Jan 28 23:03:32] DEBUG[1654]: netsock2.c:134 ast_sockaddr_split_hostport:
Splitting '192.168.2.172' into...
[Jan 28 23:03:32] DEBUG[1654]: netsock2.c:188 ast_sockaddr_split_hostport:
...host '192.168.2.172' and port ''.
[Jan 28 23:03:32] DEBUG[1654]: rtp_engine.c:345 ast_rtp_instance_new: Using
engine 'asterisk' for RTP instance '0xb7507ae8'
[Jan 28 23:03:32] DEBUG[1654]: res_rtp_asterisk.c:499 ast_rtp_new:
Allocated port 18530 for RTP instance '0xb7507ae8'
[Jan 28 23:03:32] DEBUG[1654]: rtp_engine.c:354 ast_rtp_instance_new: RTP
instance '0xb7507ae8' is setup and ready to go
[Jan 28 23:03:32] DEBUG[1654]: res_rtp_asterisk.c:2447 ast_rtp_prop_set:
Setup RTCP on RTP instance '0xb7507ae8'
  == Using SIP RTP CoS mark 5
[Jan 28 23:03:32] DEBUG[1654]: chan_sip.c:5043 do_setnat: Setting NAT on
RTP to On
[Jan 28 23:03:32] DEBUG[1654]: chan_sip.c:8816 process_sdp: Processing
session-level SDP v=0... UNSUPPORTED.
[Jan 28 23:03:32] DEBUG[1654]: chan_sip.c:8816 process_sdp: Processing
session-level SDP o=user1 2084476266 1052354777 IN IP4 192.168.2.159...
UNSUPPORTED.
[Jan 28 23:03:32] DEBUG[1654]: chan_sip.c:8816 process_sdp: Processing
session-level SDP s=-... UNSUPPORTED.
[Jan 28 23:03:32] DEBUG[1654]: netsock2.c:134 ast_sockaddr_split_hostport:
Splitting '192.168.2.159' into...
[Jan 28 23:03:32] DEBUG[1654]: netsock2.c:188 ast_sockaddr_split_hostport:
...host '192.168.2.159' and port ''.
[Jan 28 23:03:32] DEBUG[1654]: chan_sip.c:8816 process_sdp: Processing
session-level SDP c=IN IP4 192.168.2.159... OK.
[Jan 28 23:03:32] DEBUG[1654]: chan_sip.c:8816 process_sdp: Processing
session-level SDP t=0 0... UNSUPPORTED.
[Jan 28 23:03:32] WARNING[1654]: chan_sip.c:8942 process_sdp: Unsupported
SDP media type in offer: audio 0 RTP/AVP 0 8 101
[Jan 28 23:03:32] WARNING[1654]: chan_sip.c:9029 process_sdp: Failing due
to no acceptable offer found
[Jan 28 23:03:32] DEBUG[1654]: chan_sip.c:3328 __sip_xmit: Trying to put
'SIP/2.0 488' onto UDP socket destined for 192.168.2.159:5062
[Jan 28 23:03:32] DEBUG[1654]: chan_sip.c:22416 handle_request_invite: No
compatible codecs for this SIP call.
[Jan 28 23:03:32] DEBUG[1654]: chan_sip.c:25172 handle_request_do: SIP
message could not be handled, bad request:
hDVA1Kyx-1327766611250 at lucidesktop.lan
[Jan 28 23:03:32] DEBUG[1654]: chan_sip.c:8014 find_call: = Looking for
 Call ID: hDVA1Kyx-1327766611250 at lucidesktop.lan (Checking From) --From tag
grUqFtoE --To-tag as5d163902
[Jan 28 23:03:32] DEBUG[1654]: chan_sip.c:24907 handle_incoming: ****
Received ACK (6) - Command in SIP ACK
[Jan 28 23:03:32] DEBUG[1654]: chan_sip.c:4013 __sip_ack: Stopping
retransmission on 'hDVA1Kyx-1327766611250 at lucidesktop.lan' of
Response 6:
Match Found
[Jan 28 23:04:04] DEBUG[1654]: chan_sip.c:3869 __sip_autodestruct: Auto
destroying SIP dialog 'hDVA1Kyx-1327766611250 at lucidesktop.lan'
[Jan 28 23:04:04] DEBUG[1654]: chan_sip.c:6010 sip_destroy: Destroying SIP
dialog hDVA1Kyx-1327766611250 at lucidesktop.lan
[Jan 28 23:04:04] DEBUG[1654]: rtp_engine.c:293 instance_destructor:
Destroyed RTP instance '0xb7507ae8'



-
Syeh Abidin
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Kevin P. Fleming

2012-Jan-30 12:31 UTC

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[asterisk-users] process_sdp: Unsupported SDP media type in offer: audio , Failing due to no acceptable offer found

On 01/28/2012 10:22 AM, Din Assegaf wrote:> Hi All,
>
> I'm trying to upgrade asterisk server to 1.8.x from my asterisk 1.6,
>
> But when making A Call from SIP Client, I got cli Warning ... and no
> call has been made.
>
> My Sip Client is using lib java peers client http://peers.sourceforge.net/
> with standard codec PCMU/PCMA
>
> [Jan 28 23:03:32] WARNING[1654]: chan_sip.c:8942 process_sdp:
> Unsupported SDP media type in offer: audio 0 RTP/AVP 0 8 101
> [Jan 28 23:03:32] WARNING[1654]: chan_sip.c:9029 process_sdp: Failing
> due to no acceptable offer found
>
> the strange thing is when using asterisk 1.6, is normal,
> when using asterisk 1.8.x and using another client like Ekiga is normal
too,
The error message is misleading; you are having this problem because the 
'm' line in the SDP with the 'audio' offer has a port number of
0
(zero)., which means it is not an active media stream offer. It does not 
make any sense for the SDP in an INVITE for a new call to have an m-line 
with a port number of zero.

I'll improve the error message so that this sort of situation won't be 
as confusing in the future.

-- 
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kfleming at digium.com | SIP: kpfleming at digium.com | Skype: kpfleming
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at www.digium.com & www.asterisk.org

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