[Freeswitch-users] SPA921 Problem Bad Session Description

[Maxim Balabaev]

Problem was that fields with some codecs names were blank. Thats why there
were "a=rtpmap:0 /8000". Also FS don`t have aliase betwen PCMA and G711.a
for ex.
>> I'm not sure, but 5.1.8 firmware on linksys is quite outdated, isn't it?
It`s really old but this is last one on their website.

2010/10/19 Brian West <brian at freeswitch.org>

> The SDP is invalid... I suspect your PAP2 has a properly formatted SDP...
> Can you not see on the two lines for "a=rtpmap:8 /8000" and "a=rtpmap:0
> /8000" are the invalid lines they should be "a=rtpmap:8 PCMA/8000" and
> "a=rtpmap:0 PCMU/8000" .  We WILL NOT make this work because its invalid.
>  Its on you to contact the manufacture of your device and have them fix it.
>
> /b
>
> On Oct 18, 2010, at 5:32 AM, Maxim Balabaev 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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