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ffacilities
Beginner

IAD 243x tearing down SIP<>ISDN call when codec changes

I have an IAD 2431 acting as a SIP<>ISDN gateway. I've got it set up so that SIP<>ISDN calls use iLBC on the VoIP side; however, when the VoIP side puts the call on hold, connecting it to our (G.711-only) Music On Hold server, the IAD tears the call down. I've turned on 'debug ccsip media' and grabbed the following log output:

Oct 24 11:32:11.750: //1326/44A678F2800B/SIP/Media/ccsip_api_request_offer: Call has been put on hold
Oct 24 11:32:11.758: %HPI-3-FAILED_START: channel:1/0:15 DSP ID:0x2, failed mode 1 for service 27
-Traceback= 62AA72D4z 6202D9B4z 6202DEECz 62A8ACD4z 62A8B1ECz 607A1634z 607A1618z
Oct 24 11:32:11.826: //1326/44A678F2800B/SIP/Media/sipSPISetMediaSrcAddr: Media src addr for stream 1 = 1.2.3.4

Oct 24 11:32:11.826: //1326/44A678F2800B/SIP/Media/sipSPIDoPtimeNegotiation: Offered ptime:20, Negotiated ptime:20            Negotiated codec bytes: 160 for codec g711alaw
Oct 24 11:32:11.826: //1326/44A678F2800B/SIP/Media/sipSPICompareStreams: stream 1 dest_port: old=27584 new=27584
Oct 24 11:32:11.826: //1326/44A678F2800B/SIP/Media/sipSPIGetNewLocalMediaDirection:
        New Remote Media Direction = SENDRECV
        Present Local Media Direction = RECVONLY
        New Local Media Direction = SENDRECV
        retVal = 1

Oct 24 11:32:11.830: //1326/44A678F2800B/SIP/Media/sipSPIUpdateRtpSession: stun is disabled for stream:68F81370
Oct 24 11:32:11.830: //1326/44A678F2800B/SIP/Media/sipSPIUpdateRtpSession: stun is disabled for stream:68F81370
Oct 24 11:32:11.830: //1326/44A678F2800B/SIP/Media/sipSPICompareConnectionAddress: Call hold deactivated for stream            1
Oct 24 11:32:11.830: //1326/44A678F2800B/SIP/Media/sipSPICompareStreams: negotiated codec changed from ilbc to g711           alaw
Oct 24 11:32:11.830: //1326/44A678F2800B/SIP/Media/sipSPICompareStreams: Flags set for stream 1: RTP_CHANGE=Yes CAP           S_CHANGE=Yes RSVP_ADDR_CHANGE=No RSVP_MEDIA_CHANGE=Yes
Oct 24 11:32:11.830: //1326/44A678F2800B/SIP/Media/sipSPICompareSDP: Flags set for call: NEW_MEDIA=Yes DSPDNLD_REQD           =Yes IPIP_MEDIA=Yes
Oct 24 11:32:11.830: //1326/44A678F2800B/SIP/Media/sipSPIRSVPCompareSDP: Media Codec Change Identified
Oct 24 11:32:11.830: //1326/44A678F2800B/SIP/Media/sipSPIRSVPCompareSDP: Media Offer Changed
Oct 24 11:32:11.830: //1326/44A678F2800B/SIP/Media/sipSPIReplaceSDP: Main stream got changed & it's Flow Around
Oct 24 11:32:11.830: //1326/44A678F2800B/SIP/Media/sipSPIGetNewLocalMediaDirection:
        New Remote Media Direction = SENDRECV
        Present Local Media Direction = SENDRECV
        New Local Media Direction = SENDRECV
        retVal = 0

Oct 24 11:32:11.834: //1326/44A678F2800B/SIP/Media/ccsip_api_request_offer: Call has been taken off hold
Oct 24 11:32:11.834: //1326/44A678F2800B/SIP/Media/sipSPIDeleteStream: Deleting stream 1 from the VOIP RTP library
Oct 24 11:32:11.834: //1326/44A678F2800B/SIP/Media/sipSPIDestroyRtpSession: stream:68E89CFC

I assume it's line 2 (%HPI-3-FAILED_START) that's the problem here - does this mean I have a misconfiguration somewhere, or is this just broken?

Thanks!

1 REPLY 1
paolo bevilacqua
Hall of Fame Master

Can be a bug in codec re-negotiation, in which case you will have to contact the TAC.

Or just use G.711 for things to work stably.

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