Get error "415 Unsupported Media Type" when receiving call from softswitch
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@xxxx:5060;user=phone SIP/2.0 Contact: <sip:xxxx@xxxx:5061> MIME-version: 1.0 Content-Type: multipart/mixed;boundary=Alcatel-boundary To: <sip:xxxx@xxxx:5060;user=phone> From: xxxx<sip:xxxx@xxxxx:5061;user=phone>;tag=73D332463135364195291201 P-Asserted-Identity: <sip:xxxx@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@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@xxxx:5061;user=phone>;tag=73D332463135364195291201 To: <sip:xxxxx@xxxxx:5060;user=phone>;tag=m6XS44BKF4pSS Call-ID: 01FD034872814000000230A1@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@gmail.com
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@MyIP:5060;user=phone SIP/2.0 (CSeq 1) nta: canonizing sip:AAA@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@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@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@xxxx:5060;user=phone SIP/2.0 Contact: <sip:xxxx@xxxx:5061> MIME-version: 1.0 Content-Type: multipart/mixed;boundary=Alcatel-boundary To: <sip:xxxx@xxxx:5060;user=phone> From: xxxx<sip:xxxx@xxxxx:5061;user=phone>;tag=73D332463135364195291201 P-Asserted-Identity: <sip:xxxx@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@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@xxxx:5061;user=phone>;tag=73D332463135364195291201 To: <sip:xxxxx@xxxxx:5060;user=phone>;tag=m6XS44BKF4pSS Call-ID: 01FD034872814000000230A1@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@gmail.com
-- Lei.Tang lei.tlfly@gmail.com
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Lei Tang