[Linphone-developers] linphone abnormal exit and fix proposal

[Laurent Brard]

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=z9hG4bK48de2fe45128c06b2fac6af997795bced5c26a551e9a8a9797bf76ff1b42ef4e

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
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