Yep, no problem.

Here it is. Same endpoints. The only thing that changed is the call direction.

Received Message :

SIP/2.0 200 OK

Via: SIP/2.0/UDP A.B.C.D;rport;branch=z9hG4bK5Qtv3B04eS46g

From: "Test 2" <sip:xxxxxxxx75@A.B.C.D>;tag=yy2XQKXSy7gtN

To: <sip:xxxxx02@I.J.K.L>;tag=93BEF648-1A8B

Date: Tue, 24 May 2011 16:37:23 GMT

Call-ID: b7967af6-00c4-122f-19ba-000c29c18d38

CSeq: 12790979 INVITE

Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER

Allow-Events: telephone-event

Remote-Party-ID: <sip:xxxxx02@I.J.K.L>;party=called;screen=no;privacy=off

Contact: <sip:xxxxx02@I.J.K.L:5060>

Supported: replaces

Supported: sdp-anat

Server: Cisco-SIPGateway/IOS-12.x

Content-Type: application/sdp

Content-Disposition: session;handling=required

Content-Length: 334

v=0

o=CiscoSystemsSIP-GW-UserAgent 6942 3511 IN IP4 I.J.K.L

s=SIP Call

c=IN IP4 I.J.K.L

t=0 0

m=audio 17046 RTP/AVP 18 19 101 100

c=IN IP4 I.J.K.L

a=rtpmap:18 G729/8000

a=rtpmap:19 CN/8000

a=fmtp:18 annexb=no

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=rtpmap:100 X-NSE/8000

a=fmtp:100 192-194

Sent Message :

SIP/2.0 200 OK

Via: SIP/2.0/UDP E.F.G.H:5060;branch=z9hG4bK9E31DBF

From: "Test 2" <sip:xxxxxxxx75@A.B.C.D>;tag=65BCD9B0-115E

To: <sip:xxxxx02@A.B.C.D>;tag=XN94Nrcp1yt7S

Call-ID: A4A3C9E0-855A11E0-8237C79A-B2A2FA42@E.F.G.H

CSeq: 102 INVITE

Contact: <sip:xxxxx02@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: 333

Remote-Party-ID: "xxxxx02" <sip:xxxxx02@A.B.C.D>;party=calling;privacy=off;screen=no

v=0

o=FreeSWITCH 3912792884 3912792885 IN IP4 A.B.C.D

s=FreeSWITCH

c=IN IP4 A.B.C.D

t=0 0

m=audio 27728 RTP/AVP 18 100 19 101

c=IN IP4 A.B.C.D

a=rtpmap:18 G729/8000

a=fmtp:18 annexb=no

a=rtpmap:100 X-NSE/8000

a=fmtp:100 192-194

a=rtpmap:19 CN/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of David Ponzone
Sent: Monday, May 23, 2011 09:42
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] rtpmap line missing on answer

Sean,

can you show us the same packets, but for an incoming call (that works, if I understood you correctly) ?

David Ponzone Direction Technique

email: david.ponzone@ipeva.fr

tel: 01 74 03 18 97

gsm: 06 66 98 76 34

Service Client IPeva

tel: 0811 46 26 26

www.ipeva.fr - www.ipeva-studio.com

Ce message et toutes les pièces jointes sont confidentiels et établis à l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autorisée est interdite. Tout message électronique est susceptible d'altération. IPeva décline toute responsabilité au titre de ce message s'il a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce message, merci de le détruire immédiatement et d'avertir l'expéditeur.

Le 24/05/2011 à 01:53, Sean Eichhorn a écrit :

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.

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