Re: [Linphone-developers] linphone abnormal exit and fix proposal

[Simon Morlat]

Hi,

Thanks a lot.
I've just slithly modified your patch to use an enum instead of 0,1,2.

Simon

Le Jeudi 1 Décembre 2005 12:23, Laurent Brard a écrit :
> Hi,
>
>      I am testing linphone with various SIP phones and it exits when
> calling or called by a PABX/Gateway that
> sends the "telephone-event" payload first in the sdp descriptor:
>
>  - Calling:
>
> INVITE sip:address@hidden:5060 SIP/2.0
> Record-Route: <sip:10.0.0.132;lr=1>
> Supported: replaces,100rel
> User-Agent:  ABS GW v5.1.0
> To: " " <sip:address@hidden>
> From: "Poste 7132"
> <sip:address@hidden>;tag=9e06425b2f3fc5d66e0ef93de2815d64
> Contact: <sip:10.0.0.132:6060;transport=tcp>
> Content-Type: application/sdp
> Call-ID: address@hidden
> CSeq: 529507970 INVITE
> Via: SIP/2.0/udp
> 10.0.0.132;branch=z9hG4bK48de2fe45128c06b2fac6af997795bced5c26a551e9a8a9
> 797bf76ff1b42ef4e
> Via: SIP/2.0/tcp
> 10.0.0.132:6060;branch=z9hG4bK208c26be3339d28d29ff9e6a1a76767d
> Max-Forwards: 69
> Content-Length: 235
> Session-Expires: 1800
>
> v=0
> o=default 1133434445 1133434445 IN IP4 10.0.0.132
> s=abs
> c=IN IP4 10.0.0.133
> t=0 0
> m=audio 32712 RTP/AVP 97 4 8 0
> a=sendrecv
> a=rtpmap:97 telephone-event/8000
> a=rtpmap:4 G723/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:0 PCMU/8000
>
>
>  - Called:
>
> SIP/2.0 200 OK
> Record-Route: <sip:10.0.0.132;lr=1>
> Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,SUBSCRIBE,OPTIONS
> Contact: <sip:10.0.0.132:6060;transport=tcp>
> Supported: replaces,100rel
> User-Agent:  ABS GW v5.1.0
> Content-Type: application/sdp
> To: <sip:address@hidden>;tag=6bc77b029f61df88040b218b75fdf50d
> From: <sip:address@hidden>;tag=2041724643
> Call-ID: address@hidden
> CSeq: 20 INVITE
> Via: SIP/2.0/UDP 10.0.0.136:5060;rport;branch=z9hG4bK1187813012
> Content-Length: 200
>
> v=0
> o=default 1133434572 1133434573 IN IP4 10.0.0.132
> s=A conversation
> c=IN IP4 10.0.0.133
> t=0 0
> m=audio 32720 RTP/AVP 101 8
> a=rtpmap:101 telephone-event/8000
> a=rtpmap:8 PCMA/8000
> a=sendrecv
>
> Looking at RFC 2327 page 20, it says: "
> When a list of payload formats is given, this implies that all of
> these formats may be used in the session, but the first of these
> formats is the default format for the session."
>
> So it looks like the PABX/gateway should not do this. Still, linphone
> exits while it should either accept or reject the message.
> So I have changed linphone_payload_is_supported,
> linphone_read_audio_answer and linphone_read_audio_answer in exeevents.c
> to skip the "telephone-event" payload when looking for a codec:
>
> diff exeevents-origin.c linphone/coreapi/exevents.c
>
> 477,479c477
> <         /* return 1 if supported and valid codec,
> <                      2 if supported but not a valid codec (telephone
> events) */
> <               return
> (strcmp(rtppayload->mime_type,"telephone-event")!=0) ? 1 : 2;
> ---
>
> >               return 1;
>
> 503,506d500
> <     /* if payload is supported but not a valid codec, skip it */
> <     if (supported == 2) {
> <         return 0;
> <     }
> 570,573d563
> <     /* if payload is supported but not a valid codec, skip it */
> <     if (supported == 2) {
> <         return 0;
> <     }
>
> This might not be the best fix, but I think Linphone should better
> handle this case (other implementations do).
> Laurent Brard
> MWS R&D
> *            +33 2 47 74 34 06
> 7             +33 2 47 74 34 09