[Freeswitch-users] SPA921 Problem Bad Session Description
Kristian Kielhofner
kris at kriskinc.com
Tue Oct 19 07:23:26 PDT 2010
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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