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Summary:ASTERISK-07545: Problems with receiving INVITE without SDP
Reporter:Tomas Komarek (komoush)Labels:
Date Opened:2006-08-17 17:54:32Date Closed:2011-06-07 14:03:21
Priority:MajorRegression?No
Status:Closed/CompleteComponents:Channels/chan_sip/Interoperability
Versions:Frequency of
Occurrence
Related
Issues:
Environment:Attachments:( 0) debug_with_sip.txt
Description:I have encountered a problem with INVITE from UAC without the SDP part. I use asterisk on a FreeBSD 5.4.

Asterisk must offer a codec to use in 200OK if it does not receive a SDP in the INVITE, but asterist does not offer any. Here is the asterisk log:

   -- Executing Dial("SIP/anonymous.invalid-086d7000", "SIP/226549022|5|t") in new stack
Aug 18 00:50:38 WARNING[845]: channel.c:2494 ast_request: Channel SIP does not support requested formats ()
Aug 18 00:50:38 NOTICE[845]: app_dial.c:1040 dial_exec_full: Unable to create channel of type 'SIP' (cause 0 - Unknown)
 == Everyone is busy/congested at this time (1:0/0/1)
   -- Executing VoiceMail("SIP/anonymous.invalid-086d7000", "u226549022") in new stack
Aug 18 00:50:38 WARNING[845]: channel.c:2320 set_format: Unable to find a codec translation path from 0x0 (nothing) to gsm
Aug 18 00:50:38 WARNING[845]: translate.c:88 powerof: Powerof 0: No power??
Aug 18 00:50:38 WARNING[845]: translate.c:88 powerof: Powerof 0: No power??
Aug 18 00:50:38 WARNING[845]: translate.c:133 ast_translator_build_path: No translator path from g723 to unknown
Aug 18 00:50:38 WARNING[845]: chan_sip.c:2560 sip_write: Asked to transmit frame type 2, while native formats is 0 (read/write = 0/2)
   -- Playing '/var/spool/asterisk/voicemail/default/226549022/unavail' (language 'en')
 == Spawn extension (default, 226549022, 2) exited non-zero on 'SIP/anonymous.invalid-086d7000'


****** ADDITIONAL INFORMATION ******

RFC3261: 13.2.1 Creating the Initial INVITE

The initial offer MUST be in either an INVITE or, if not there,
in the first reliable non-failure message from the UAS back to
the UAC.  In this specification, that is the final 2xx
response.

If the initial offer is in an INVITE, the answer MUST be in a
reliable non-failure message from UAS back to UAC which is
correlated to that INVITE.  For this specification, that is
only the final 2xx response to that INVITE.  That same exact
answer MAY also be placed in any provisional responses sent
prior to the answer.  The UAC MUST treat the first session
description it receives as the answer, and MUST ignore any
session descriptions in subsequent responses to the initial
INVITE.

If the initial offer is in the first reliable non-failure
message from the UAS back to UAC, the answer MUST be in the
acknowledgement for that message (in this specification, ACK
for a 2xx response).

Comments:By: Serge Vecher (serge-v) 2006-08-18 08:09:53

As per bug guidelines, you need to attach a SIP debug trace illustrating the problem. Please do the following:
1) Prepare test environment (reduce the ammount of unrelated traffic on the server);
2) Make sure your logger.conf has the following line:
  console => notice,warning,error,debug
3) restart Asterik.
4) Enable SIP transaction logging with the following CLI commands:
set debug 4
set verbose 4
sip debug
5) Save complete console log to file and _attach_ said file to the bug.
By: Tomas Komarek (komoush) 2006-08-18 08:52:05

The file is uploaded.
By: Serge Vecher (serge-v) 2006-08-18 09:02:42

'sip debug' command was not entered.
By: Tomas Komarek (komoush) 2006-08-18 10:01:36

Now with sip debug... Sorry...
By: Olle Johansson (oej) 2006-08-30 12:18:17

Here's the problem we need to fix:

-- Executing Dial("SIP/sip.broadnet.cz-086b5000", "SIP/226549022|5|t") in new stack
Aug 18 16:09:29 WARNING[12544]: channel.c:2494 ast_request: Channel SIP does not support requested formats ()

Can any other channel set up a call with no formats?
By: Tomas Komarek (komoush) 2006-08-30 13:30:34

Sorry, I use asterisk at this moment just as a sip peer and I don't have a chance to test this with another type of signalization. When do you think you can post a patch?
By: Olle Johansson (oej) 2006-08-30 14:09:16

That depends on when I or someone else have time to do this...
By: Tomas Komarek (komoush) 2006-08-30 14:28:48

Sounds good, thanks a lot for this in advance...
By: Tomas Komarek (komoush) 2006-09-13 02:03:02

Hello, something new in the case? Thanks for an answer.
By: Olle Johansson (oej) 2006-10-29 10:07:19.000-0600

I am not sure we do support this. If not, we should at least issue a proper error message.
By: Olle Johansson (oej) 2006-10-29 10:09:11.000-0600

CAn we see the SIP configuration for these devices? For some reason, Asterisk does send a 200 OK with SDP, but with no codecs.
By: Tomas Komarek (komoush) 2006-11-01 07:22:21.000-0600

How do you mean sSIP configuration it is a standard peer connected to the Asterisk. Configuration is hard to describe. I think most of the things can be seen in the trace...
By: Serge Vecher (serge-v) 2006-11-06 12:54:35.000-0600

komoush: by SIP configuration we mean respective entries in sip.conf sans any private information (secret, ip address).
By: Anthony LaMantia (alamantia) 2007-01-05 16:41:52.000-0600

komoush,
can you upload the requested configuration information?
By: Olle Johansson (oej) 2007-02-01 16:15:20.000-0600

No answer from reporter. Please re-open when you're ready to respond.

I believe this is currently not supported.