Hi guys, I've been using FreeSWITCH for a couple of weeks and built a PBX and B2BUA out of it. I've seen some interesting things that happen when PBXs get m= headers that they don't like (such as Asterisk balking when it gets multiple video streams), and I've seen one recently while calling an old school PBX. Here's my build: Me(Linphone) -> PBX (FreeSWITCH) -> B2BUA (FreeSWITCH) -> Destination The B2BUA uses proxy_media mode and doesn't have a problem, but the PBX uses passthrough codecs (so it's reading the SDP packet) and sees a problem, throwing a 500 error back to my softphone The outbound packet from the PBX to the B2BUA: send 1428 bytes to udp/[210.7.45.66]:5060 at 02:46:48.407768: ------------------------------------------------------------------------ INVITE sip:+61269337555@210.7.45.66 SIP/2.0 Via: SIP/2.0/UDP 210.7.46.240;rport;branch=z9hG4bKZrKQvvHNSy64m Max-Forwards: 69 From: "Sam Russell" <sip:+6442820079@210.7.46.240>;tag=e722ggarBe7je To: <sip:+61269337555@210.7.45.66> Call-ID: 1d80cd44-781c-1231-8db7-0050568a49fc CSeq: 47499340 INVITE Contact: <sip:gw+sbc@210.7.46.240:5060;transport=udp;gw=sbc> User-Agent: FreeSWITCH-mod_sofia/1.3.13b+git~20130223T185558Z~28680c5e58 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, conference, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 453 X-FS-Support: update_display,send_info Remote-Party-ID: "Sam Russell" <sip:+6442820079@210.7.46.240

;party=calling;screen=yes;privacy=off

v=0 o=FreeSWITCH 1375638308 1375638309 IN IP4 210.7.46.240 s=FreeSWITCH c=IN IP4 210.7.46.240 t=0 0 m=audio 32500 RTP/AVP 98 99 100 0 8 101 13 a=rtpmap:98 SPEEX/32000 a=fmtp:98 vbr=on a=rtpmap:99 SPEEX/16000 a=fmtp:99 vbr=on a=rtpmap:100 SPEEX/8000 a=fmtp:100 vbr=on a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 m=video 29910 RTP/AVP 102 103 a=rtpmap:102 H263-1998/90000 a=fmtp:102 CIF=1;QCIF=1 a=rtpmap:103 VP8/90000 The offending packet that it returns: ------------------------------------------------------------------------ recv 1357 bytes from udp/[210.7.45.66]:5060 at 02:46:48.739816: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 210.7.46.240;rport=5060;branch=z9hG4bKZrKQvvHNSy64m From: "Sam Russell" <sip:+6442820079@210.7.46.240>;tag=e722ggarBe7je To: <sip:+61269337555@210.7.45.66>;tag=teBFaBg5DjBHm Call-ID: 1d80cd44-781c-1231-8db7-0050568a49fc CSeq: 47499340 INVITE Contact: <sip:+61269337555@210.7.45.66:5060;transport=udp> User-Agent: FreeSWITCH-mod_sofia/1.2.11+git~20130723T222215Z~55b6b8424f Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, conference, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 242 X-FS-Display-Name: Outbound Call X-FS-Display-Number: sip:+61269337555@210.7.45.66 x-inin-crn: 1101488375;loc=%3cRegionDefaultLocation%3e X-FS-Support: update_display,send_info Remote-Party-ID: "Outbound Call" <sip:+61269337555@210.7.45.66

;party=calling;privacy=off;screen=no

v=0 o=FreeSWITCH 914114385 914114386 IN IP4 210.7.45.66 s=FreeSWITCH c=IN IP4 210.7.45.66 t=0 0 m=audio 17704 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 m=video 0 RTP/AVP 96 a=rtpmap:96 /0 The PBX then balks and sends a 500 error to my softphone send 704 bytes to udp/[210.7.45.66]:5060 at 02:46:48.740796: ------------------------------------------------------------------------ BYE sip:+61269337555@210.7.45.66:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 210.7.46.240;rport;branch=z9hG4bK1a68ZjKvKgKac Max-Forwards: 70 From: "Sam Russell" <sip:+6442820079@210.7.46.240>;tag=e722ggarBe7je To: <sip:+61269337555@210.7.45.66>;tag=teBFaBg5DjBHm Call-ID: 1d80cd44-781c-1231-8db7-0050568a49fc CSeq: 47499341 BYE Contact: <sip:gw+sbc@210.7.46.240:5060;transport=udp;gw=sbc> User-Agent: FreeSWITCH-mod_sofia/1.3.13b+git~20130223T185558Z~28680c5e58 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Reason: SIP;cause=488;text="Incomplete offer/answer" Content-Length: 0 ------------------------------------------------------------------------ Then I see this in the logs straight after 2013-08-05 14:46:48.736200 [DEBUG] sofia.c:1431 nua_i_media_error: unknown event 22: 988 Incomplete offer/answer The only thing odd about the SDP packet that I get back is the attribute after the m=video line. I think FreeSWITCH sees this, doesn't ignore the a= line like it should, and then gets upset with not being able to match anything before the "/0". RFC 4566 (5.13) says that "If an attribute is received that is not understood, it MUST be ignored by the receiver." So FreeSWITCH should be ignoring any "a=" attributes after it gets a "m=" line with a 0 port, but should also ignore the attribute if it can't match (but maybe leave a warning/notice in logs)? Does this look right? Has anyone else seen this, and is there a workaround? Cheers Sam