[Freeswitch-users] Get error "415 Unsupported Media Type" when receiving call from softswitch

Lei Tang lei.tlfly at gmail.com
Sun Nov 1 20:09:25 PST 2009


FYI, Here is the log when I set sofia loglevel all 9
==============
tport_wakeup_pri(00DFE3E8): events IN
tport_recv_event(00DFE3E8)
tport(00DFE3E8) msg 01B2E0C0 from (udp/MyIP:5060) has 1315 bytes, veclen
= 1
tport(00DFE3E8): msg 01B2E0C0 (1315 bytes) from udp/SSIP:5060/sip next=000
00000
nta: received INVITE sip:AAA at MyIP:5060;user=phone SIP/2.0 (CSeq
1)
nta: canonizing sip:AAA at MyIP:5060 with contact
nta: INVITE (1) going to a default leg
nta: timer shortened to 200 ms
tport_tsend(00DFE3E8) tpn = UDP/SSIP:5061
tport_resolve addrinfo = SSIP:5061
tport(00DFE3E8): not found by name UDP/SSIP:5061
tport_vsend(00DFE3E8): 652 bytes of 652 to udp/SSIP:5061
tport_vsend returned 652
nta: sent 415 Unsupported Media Type for INVITE (1)
tport_wakeup_pri(00DFE3E8): events IN
tport_recv_event(00DFE3E8)
tport(00DFE3E8) msg 0192C510 from (udp/MyIP:5060) has 388 bytes, veclen =
 1
tport(00DFE3E8): msg 0192C510 (388 bytes) from udp/SSIP:5060/sip next=0000
0000
nta: received ACK sip:AAA at MyIP:5060;user=phone SIP/2.0 (CSeq 1)
nta: ACK (1) is going to INVITE (1)
nta: timer set next to 937 ms
nta: timer J fired, terminate 200 response
incoming_reclaim_all(00000000, 00000000, 02E2FEB8)
nta_incoming_timer: 0/0 resent, 0/0 tout, 1/2 term, 1/2 free
nta: timer set next to 3859 ms
nta: timer I fired, terminate 415 response
incoming_reclaim_all(00000000, 00000000, 02E2FEB8)
nta_incoming_timer: 0/0 resent, 0/0 tout, 1/1 term, 1/1 free
nta: timer not set

2009/11/2 Lei Tang <lei.tlfly at gmail.com>

> Hi all, I get a  "415 Unsupported Media Type" when FS receiving call from a
> softswitch. I captured some packets, It seems that the softswitch use SIP-I
> protocol, does FS can handle SIP-I message?
>
> ===here is the invite messagefrom softswitch
> INVITE sip:xxxxx at xxxx:5060;user=phone SIP/2.0
> Contact: <sip:xxxx at xxxx:5061>
> MIME-version: 1.0
> Content-Type: multipart/mixed;boundary=Alcatel-boundary
> To: <sip:xxxx at xxxx:5060;user=phone>
> From: xxxx<sip:xxxx at xxxxx:5061;user=phone>;tag=73D332463135364195291201
> P-Asserted-Identity: <sip:xxxx at xxxxx:5061;user=phone>
> Allow:
> INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,COMET,UPDATE,PRACK,REFER,SUBSCRIBE,NOTIFY,MESSAGE
> Supported: 100rel,timer,replaces,diversion
> Expires: 155
> Session-Expires: 1800
> Min-SE: 90
> Call-ID: 01FD034872814000000230A1 at sip-3
> Max-Forwards: 70
> CSeq: 1 INVITE
> Timestamp: 10645
> Via: SIP/2.0/UDP xxxxx:5061;branch=z9hG4bK8E1558EA4F1BE09A2BCB669331A9AC7E
> Content-Length: 542
>
> --Alcatel-boundary
> Content-Type: application/sdp
>
> v=0
> o=- 5 8 IN IP4 xxxxxx
> s=SDP Data
> c=IN IP4 xxxxx
> t=0 0
> m=audio 10266 RTP/AVP 0 8 96 110 111 112 113 3 97
> a=rtpmap:110 speex/8000/1
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:96 TELEPHONE-EVENT/8000
> a=fmtp:110 mode=3
> a=rtpmap:111 speex/8000/1
> a=rtpmap:3 GSM/8000
> a=ptime:30
> a=fmtp:111 mode=2
> a=rtpmap:112 speex/8000/1
>
> --Alcatel-boundary
> Content-Type: application/ISUP;version=N/A
>
> .. ....
> ...0.D...
> ...V..0......1...n..p...
> --Alcatel-boundary--
>
> =====response message from FS
> SIP/2.0 415 Unsupported Media Type
> Via: SIP/2.0/UDP xxxxx:5061;branch=z9hG4bK8E1558EA4F1BE09A2BCB669331A9AC7E
> From: xxxxx <sip:xxxxx at xxxx:5061;user=phone>;tag=73D332463135364195291201
> To: <sip:xxxxx at xxxxx:5060;user=phone>;tag=m6XS44BKF4pSS
> Call-ID: 01FD034872814000000230A1 at sip-3
> CSeq: 1 INVITE
> Timestamp: 10645 0.000000
> User-Agent: FreeSWITCH-mod_sofia/1.0.4-14460
> Accept: application/sdp
> Accept-Encoding:
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,
> NOTIFY, REFER, UPDATE, REGISTER, INFO
> Supported: timer, precondition, path, replaces
> Allow-Events: talk, refer
> Content-Length: 0
> --
> Lei.Tang
> lei.tlfly at gmail.com
>



-- 
Lei.Tang
lei.tlfly at gmail.com
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