Yeah, they're negotiating the same codec on both sides. The problem is the way Cisco handles the negotiation. Here's an example of what I see in Freeswitch: Received from CALLED : SIP/2.0 200 OK Via: SIP/2.0/UDP A.B.C.D;rport;branch=z9hG4bKXZZ18p8tD03cS From: "pending" <sip:xxxxxxxx02@A.B.C.D>;tag=NH1HrNe744HSB To: <sip:sean@E.F.G.H:5060>;tag=622555CC-408 Date: Mon, 23 May 2011 23:50:56 GMT Call-ID: 69258f8c-0038-122f-19ba-000c29c18d38 CSeq: 12760849 INVITE Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER Allow-Events: telephone-event Contact: <sip:xxxxxxxx75@E.F.G.H:5060> Supported: replaces Supported: sdp-anat Server: Cisco-SIPGateway/IOS-12.x Content-Type: application/sdp Content-Disposition: session;handling=required Content-Length: 311 v=0 o=CiscoSystemsSIP-GW-UserAgent 4080 7999 IN IP4 E.F.G.H s=SIP Call c=IN IP4 E.F.G.H t=0 0 m=audio 17122 RTP/AVP 0 19 101 100 c=IN IP4 E.F.G.H a=rtpmap:0 PCMU/8000 a=rtpmap:19 CN/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:100 X-NSE/8000 a=fmtp:100 192-194 Sent to CALLING : SIP/2.0 200 OK Via: SIP/2.0/UDP I.J.K.L:5060;branch=z9hG4bK4C2B6C64 From: <sip:xxxxxxxx02@ I.J.K.L >;tag=9020153C-21B3 To: <sip:xxxxxxxx75@A.B.C.D>;tag=m87rptX37UU6F Call-ID: AFADEF2E-84CC11E0-B1D4F23A-27804614@ I.J.K.L CSeq: 101 INVITE Contact: <sip:xxxxxxxx75@ A.B.C.D:5060;transport=udp> User-Agent: GCI Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Min-SE: 1800 Content-Type: application/sdp Content-Disposition: session Content-Length: 261 Remote-Party-ID: "Outbound Call" <sip:sean@ A.B.C.D >;party=calling;privacy=off;screen=no v=0 o=FreeSWITCH 1995227603 1995227604 IN IP4 A.B.C.D s=FreeSWITCH c=IN IP4 A.B.C.D t=0 0 m=audio 24636 RTP/AVP 0 19 101 c=IN IP4 A.B.C.D a=rtpmap:0 PCMU/8000 a=rtpmap:19 CN/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 This is not the case when the call direction is reversed. Like I said, the only difference appears to be that one direction is responding with a 183 while this one is responding with a 200.