[Freeswitch-users] SPA921 Problem Bad Session Description

Tue Oct 19 07:37:51 PDT 2010

We already told him this when he filed a JIRA on it.
It's clearly broken and probably fixed with a firmware update to the device.

On Tue, Oct 19, 2010 at 9:23 AM, Kristian Kielhofner <kris at kriskinc.com> wrote:
> Maxim,
>
>  That SDP looks nasty:
>
>   a=rtpmap:8 /8000
>   a=rtpmap:0 /8000
>
>  These are static payload types and don't require an rtpmap line (the
> "offer" is in the m= line).  However, when you use an rtpmap you must
> use the IANA payload type names: PCMU and PCMA in this case.  Either
> of these two SDPs would be valid:
>
>   v=0
>   o=- 603544 603544 IN IP4 xxx
>   s=-
>   c=IN IP4 xxx
>   t=0 0
>   m=audio 16406 RTP/AVP 8 0 2 4 18 96 97 98 101
>   a=rtpmap:2 G726-32/8000
>   a=rtpmap:4 G723/8000
>   a=rtpmap:18 G729/8000
>   a=rtpmap:96 G726-40/8000
>   a=rtpmap:97 G726-24/8000
>   a=rtpmap:98 G726-16/8000
>   a=rtpmap:101 telephone-event/8000
>   a=fmtp:101 0-15
>   a=ptime:30
>   a=sendrecv
>
> -or-
>
>   v=0
>   o=- 603544 603544 IN IP4 xxx
>   s=-
>   c=IN IP4 xxx
>   t=0 0
>   m=audio 16406 RTP/AVP 8 0 2 4 18 96 97 98 101
>   a=rtpmap:8 PCMA/8000
>   a=rtpmap:0 PCMU/8000
>   a=rtpmap:2 G726-32/8000
>   a=rtpmap:4 G723/8000
>   a=rtpmap:18 G729/8000
>   a=rtpmap:96 G726-40/8000
>   a=rtpmap:97 G726-24/8000
>   a=rtpmap:98 G726-16/8000
>   a=rtpmap:101 telephone-event/8000
>   a=fmtp:101 0-15
>   a=ptime:30
>   a=sendrecv
>
>  Yours isn't :).
>
> http://www.iana.org/assignments/rtp-parameters
>
> On Mon, Oct 18, 2010 at 6:32 AM, Maxim Balabaev <balabaev.m at gmail.com> wrote:
>> I can`t make calls from linksys spa921 because of "400 Bad Session
>> Description", incoming are ok. pap2t works perfectly. freeswitch is rev from
>> git trunk. Here comes logs:
>>    ------------------------------------------------------------------------
>> recv 868 bytes from udp/[xxx]:5060 at 10:27:30.709397:
>>    ------------------------------------------------------------------------
>>    INVITE sip:1001 at xxx SIP/2.0
>>    Via: SIP/2.0/UDP xxx:5060;branch=z9hG4bK-ede98b83;rport
>>    From: <sip:022 at xxx>;tag=8f6dbe61c1d0552ao0
>>    To: <sip:1001 at xxx>
>>    Call-ID: 14d793ed-1abf9d0e at xxx
>>    CSeq: 101 INVITE
>>    Max-Forwards: 70
>>    Contact: <sip:022 at xxx:5060>
>>    Expires: 240
>>    User-Agent: Linksys/SPA921-5.1.8
>>    Content-Length: 386
>>    Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
>>    Supported: replaces
>>    Content-Type: application/sdp
>>
>>    v=0
>>    o=- 603544 603544 IN IP4 xxx
>>    s=-
>>    c=IN IP4 xxx
>>    t=0 0
>>    m=audio 16406 RTP/AVP 8 0 2 4 18 96 97 98 101
>>    a=rtpmap:8 /8000
>>    a=rtpmap:0 /8000
>>    a=rtpmap:2 G726-32/8000
>>    a=rtpmap:4 G723/8000
>>    a=rtpmap:18 G729/8000
>>    a=rtpmap:96 G726-40/8000
>>    a=rtpmap:97 G726-24/8000
>>    a=rtpmap:98 G726-16/8000
>>    a=rtpmap:101 telephone-event/8000
>>    a=fmtp:101 0-15
>>    a=ptime:30
>>    a=sendrecv
>>    ------------------------------------------------------------------------
>> send 562 bytes to udp/[xxx]:5060 at 10:27:30.709607:
>>    ------------------------------------------------------------------------
>>    SIP/2.0 400 Bad Session Description
>>    Via: SIP/2.0/UDP xxx:5060;branch=z9hG4bK-ede98b83;rport=5060
>>    From: <sip:022 at xxx>;tag=8f6dbe61c1d0552ao0
>>    To: <sip:1001 at xxx>;tag=ytUD271ypvy6r
>>    Call-ID: 14d793ed-1abf9d0e at xxx
>>    CSeq: 101 INVITE
>>    User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-f8d8a39 2010-10-18 03-19-16
>> -0400
>>    Accept: application/sdp
>>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
>> REGISTER, REFER, NOTIFY
>>    Supported: timer, precondition, path, replaces
>>    Allow-Events: talk, hold, refer
>>    Content-Length: 0
>>
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>>
>
>
>
> --
> Kristian Kielhofner
> http://www.astlinux.org
> http://blog.krisk.org
> http://www.star2star.com
> http://www.submityoursip.com
> http://www.voalte.com
>
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