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488 Invalid incoming Gateway SDP: Invalid media RRS feed

  • Question

  • We're currently performing interop tests beteen OCS R2 and other PBXes in our lab (various releases of Cisco Call Manager and Alcatel-Lucent Omni PCX Enterprise).

    We've run across an interesting issue where a call initiated by the Alcatel System is rejected with a 488 status code and an Invalid Gateway media error message.

    The whole calls setup looks as follows:

    Actors: caller: 3462 on PBX 10.145.27.72
    Mediation Server: 10.145.206.156

    INVITE sip:9000@10.145.206.156;transport=TCP;user=phone SIP/2.0
    Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, SUBSCRIBE, OPTIONS, UPDATE, INFO
    Supported: replaces,timer,100rel
    User-Agent: OmniPCX Enterprise R9.0 h1.301.25
    Session-Expires: 1800;refresher=uac
    Min-SE: 900
    P-Asserted-Identity: "Tarzis Kessler" <sip:3462@10.145.27.72;user=phone>
    Content-Type: application/sdp
    To: <sip:9000@10.145.206.156;user=phone>
    From: "Tarzis Kessler" <sip:3462@10.145.27.72>;tag=a46f56845e2649610af4db5cbb5864ba
    Contact: <sip:10.145.27.72;transport=TCP>
    Call-ID: 01b1aa944c20bcad404dfd3e4cdc8b30@10.145.27.72
    CSeq: 1993218136 INVITE
    Via: SIP/2.0/TCP 10.145.27.72;branch=z9hG4bK1b582e17fe997251482147f190c49344
    Max-Forwards: 70
    Content-Length: 316

    v=0
    o=OXE 1233842864 1233842864 IN IP4 10.145.27.72
    s=abs
    c=IN IP4 10.145.26.27
    t=0 0
    m=audio 32514 RTP/AVP 8 0 4 97
    a=sendrecv
    a=rtpmap:8 PCMA/8000
    a=ptime:20
    a=maxptime:30
    a=rtpmap:0 PCMU/8000
    a=ptime:20
    a=maxptime:30
    a=rtpmap:4 G723/8000
    a=ptime:30
    a=maxptime:30
    a=rtpmap:97 telephone-event/8000

    SIP/2.0 100 Trying
    FROM: "Tarzis Kessler"<sip:3462@10.145.27.72>;tag=a46f56845e2649610af4db5cbb5864ba
    TO: <sip:9000@10.145.206.156;user=phone>
    CSEQ: 1993218136 INVITE
    CALL-ID: 01b1aa944c20bcad404dfd3e4cdc8b30@10.145.27.72
    VIA: SIP/2.0/TCP 10.145.27.72;branch=z9hG4bK1b582e17fe997251482147f190c49344
    CONTENT-LENGTH: 0

    SIP/2.0 488 Invalid incoming Gateway SDP: Invalid media
    FROM: "Tarzis Kessler"<sip:3462@10.145.27.72>;tag=a46f56845e2649610af4db5cbb5864ba
    TO: <sip:9000@10.145.206.156;user=phone>;epid=3CAD90C5CE;tag=43d88de011
    CSEQ: 1993218136 INVITE
    CALL-ID: 01b1aa944c20bcad404dfd3e4cdc8b30@10.145.27.72
    VIA: SIP/2.0/TCP 10.145.27.72;branch=z9hG4bK1b582e17fe997251482147f190c49344
    CONTENT-LENGTH: 0
    SERVER: RTCC/3.5.0.0 MediationServer

    I sniffed the entire traffic and there's no a single RTP packet going either way so it's really something about the INVITE that the mediation server doesn't like. Does anybody have an idea what that could be? The Contact field seems a little suspicious but the user part may be left out as it seems, but other than that it looks pretty standard to me.

    Regards
    Stephan
    Friday, February 6, 2009 11:05 AM

All replies

  • My colleagues asked me to do some more detailed traces which reveal the culprit:

    $$END-MEDIATIONSERVER
    TL_WARN(TF_COMPONENT) [1]07E4.02FC::02/12/2009-13:23:27.875.0000017f (MediationServer,MCCapabilitySet.CreateAudioMCMediaFormatFromSdp:mctypes.cs(272))Unexpected codecID CodecG7231 not found in the global dictionary. Codec name: G723/8000
    TL_ERROR(TF_PROTOCOL) [1]07E4.02FC::02/12/2009-13:23:27.875.00000180 (MediationServer,GatewaySDP.ParseSdpOffer:gatewaysdp.cs(1207))( 0000000001B26EB8 )$$START-MEDIATIONSERVER
    MediationCall: 16fda7ebf94a439ab9082525ff225b0c
    CallId: 40ef70b2c54db65e2cb0754445efe07a@10.145.27.72
    From: sip:+41448153462@10.145.27.72
    To: sip:+419000@10.145.206.156;user=phone
    Direction: Inbound
    Start-Line: SDP parsing failed: Invalid media

    Regardless of whether a codec is supported or not, this shouldn't happen - a sip device is supposed to ignore codecs it doesn't support and respond with the one it picked. I suppose the question becomes if it's possible to simply add the codec to the global dictionary.
    Thursday, February 12, 2009 4:58 PM