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