/*! \brief Codecs that we support by default: */
static int global_capability = AST_FORMAT_ULAW | AST_FORMAT_ALAW | AST_FORMAT_GSM | AST_FORMAT_H263;
-static int noncodeccapability = AST_RTP_DTMF;
/* Object counters */
static int suserobjs = 0; /*!< Static users */
int peercapability; /*!< Supported peer capability */
int prefcodec; /*!< Preferred codec (outbound only) */
int noncodeccapability; /*!< DTMF RFC2833 telephony-event */
+ int jointnoncodeccapability; /*!< Joint Non codec capability */
int redircodecs; /*!< Redirect codecs */
int maxcallbitrate; /*!< Maximum Call Bitrate for Video Calls */
struct t38properties t38; /*!< T38 settings */
newjointcapability = p->capability & (peercapability | vpeercapability);
newpeercapability = (peercapability | vpeercapability);
- newnoncodeccapability = noncodeccapability & peernoncodeccapability;
+ newnoncodeccapability = p->noncodeccapability & peernoncodeccapability;
if (debug) {
ast_getformatname_multiple(s4, BUFSIZ, newjointcapability));
ast_verbose("Non-codec capabilities (dtmf): us - %s, peer - %s, combined - %s\n",
- ast_rtp_lookup_mime_multiple(s1, BUFSIZ, noncodeccapability, 0, 0),
+ ast_rtp_lookup_mime_multiple(s1, BUFSIZ, p->noncodeccapability, 0, 0),
ast_rtp_lookup_mime_multiple(s2, BUFSIZ, peernoncodeccapability, 0, 0),
ast_rtp_lookup_mime_multiple(s3, BUFSIZ, newnoncodeccapability, 0, 0));
}
/* We are now ready to change the sip session and p->rtp and p->vrtp with the offered codecs, since
they are acceptable */
- p->jointcapability = newjointcapability; /* Our joint codec profile for this call */
- p->peercapability = newpeercapability; /* The other sides capability in latest offer */
- p->noncodeccapability = newnoncodeccapability; /* DTMF capabilities */
+ p->jointcapability = newjointcapability; /* Our joint codec profile for this call */
+ p->peercapability = newpeercapability; /* The other sides capability in latest offer */
+ p->jointnoncodeccapability = newnoncodeccapability; /* DTMF capabilities */
ast_rtp_pt_copy(p->rtp, newaudiortp);
if (p->vrtp)
/* Now add DTMF RFC2833 telephony-event as a codec */
for (x = 1; x <= AST_RTP_MAX; x <<= 1) {
- if (!(p->noncodeccapability & x))
+ if (!(p->jointnoncodeccapability & x))
continue;
add_noncodec_to_sdp(p, x, 8000,