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Merged revisions 229912 via svnmerge from

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https://origsvn.digium.com/svn/asterisk/trunk

........
  r229912 | file | 2009-11-13 09:56:16 -0600 (Fri, 13 Nov 2009) | 2 lines
  
  Fix T.38 negotiation regression introduced with the SDP parser changes.
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@229915 65c4cc65-6c06-0410-ace0-fbb531ad65f3
1.6.2
Joshua Colp 16 years ago
parent 6066aadc52
commit 1341d8df05
1 changed files with 56 additions and 51 deletions
Show Stats Download Patch File Download Diff File

107
channels/chan_sip.c
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@ -8015,9 +8015,11 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
struct ast_hostent audiohp; struct ast_hostent audiohp;
struct ast_hostent videohp; struct ast_hostent videohp;
struct ast_hostent texthp; struct ast_hostent texthp;
struct ast_hostent imagehp;
struct hostent *hp = NULL; /*!< RTP Audio host IP */ struct hostent *hp = NULL; /*!< RTP Audio host IP */
struct hostent *vhp = NULL; /*!< RTP video host IP */ struct hostent *vhp = NULL; /*!< RTP video host IP */
struct hostent *thp = NULL; /*!< RTP text host IP */ struct hostent *thp = NULL; /*!< RTP text host IP */
struct hostent *ihp = NULL; /*!< UDPTL host ip */
int portno = -1; /*!< RTP Audio port number */ int portno = -1; /*!< RTP Audio port number */
int vportno = -1; /*!< RTP Video port number */ int vportno = -1; /*!< RTP Video port number */
int tportno = -1; /*!< RTP Text port number */ int tportno = -1; /*!< RTP Text port number */
@ -8126,6 +8128,7 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
hp = &sessionhp.hp; hp = &sessionhp.hp;
vhp = hp; vhp = hp;
thp = hp; thp = hp;
ihp = hp;
} }
break; break;
case 'a': case 'a':
@ -8240,15 +8243,6 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
if (p->t38.state != T38_ENABLED) { if (p->t38.state != T38_ENABLED) {
memset(&p->t38.their_parms, 0, sizeof(p->t38.their_parms)); memset(&p->t38.their_parms, 0, sizeof(p->t38.their_parms));
/* Remote party offers T38, we need to update state */
if ((t38action == SDP_T38_ACCEPT) &&
(p->t38.state == T38_LOCAL_REINVITE)) {
change_t38_state(p, T38_ENABLED);
} else if ((t38action == SDP_T38_INITIATE) &&
p->owner && p->lastinvite) {
change_t38_state(p, T38_PEER_REINVITE); /* T38 Offered in re-invite from remote party */
}
} }
} else { } else {
ast_log(LOG_WARNING, "Unsupported SDP media type in offer: %s\n", m); ast_log(LOG_WARNING, "Unsupported SDP media type in offer: %s\n", m);
@ -8282,6 +8276,11 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
processed = TRUE; processed = TRUE;
thp = &texthp.hp; thp = &texthp.hp;
} }
} else if (image) {
if (process_sdp_c(value, &imagehp)) {
processed = TRUE;
ihp = &imagehp.hp;
}
} }
break; break;
case 'a': case 'a':
@ -8366,48 +8365,10 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
ast_rtp_lookup_mime_multiple(s2, SIPBUFSIZE, peernoncodeccapability, 0, 0), ast_rtp_lookup_mime_multiple(s2, SIPBUFSIZE, peernoncodeccapability, 0, 0),
ast_rtp_lookup_mime_multiple(s3, SIPBUFSIZE, newnoncodeccapability, 0, 0)); ast_rtp_lookup_mime_multiple(s3, SIPBUFSIZE, newnoncodeccapability, 0, 0));
} }
if (!newjointcapability) { if (!newjointcapability && (portno != -1)) {
/* If T.38 was not negotiated either, totally bail out... */ ast_log(LOG_NOTICE, "No compatible codecs, not accepting this offer!\n");
if ((p->t38.state == T38_DISABLED) || !udptlportno) { /* Do NOT Change current setting */
ast_log(LOG_NOTICE, "No compatible codecs, not accepting this offer!\n"); return -1;
/* Do NOT Change current setting */
return -1;
} else {
ast_debug(3, "Have T.38 but no audio codecs, accepting offer anyway\n");
return 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->jointnoncodeccapability = newnoncodeccapability; /* DTMF capabilities */
if (p->trtp && (p->jointcapability & AST_FORMAT_T140RED)) {
p->red = 1;
rtp_red_init(p->trtp, 300, red_data_pt, 2);
} else {
p->red = 0;
}
ast_rtp_pt_copy(p->rtp, newaudiortp);
if (p->vrtp)
ast_rtp_pt_copy(p->vrtp, newvideortp);
if (p->trtp)
ast_rtp_pt_copy(p->trtp, newtextrtp);
if (ast_test_flag(&p->flags[0], SIP_DTMF) == SIP_DTMF_AUTO) {
ast_clear_flag(&p->flags[0], SIP_DTMF);
if (newnoncodeccapability & AST_RTP_DTMF) {
/* XXX Would it be reasonable to drop the DSP at this point? XXX */
ast_set_flag(&p->flags[0], SIP_DTMF_RFC2833);
/* Since RFC2833 is now negotiated we need to change some properties of the RTP stream */
ast_rtp_setdtmf(p->rtp, 1);
ast_rtp_setdtmfcompensate(p->rtp, ast_test_flag(&p->flags[1], SIP_PAGE2_RFC2833_COMPENSATE));
} else {
ast_set_flag(&p->flags[0], SIP_DTMF_INBAND);
}
} }
/* Setup audio address and port */ /* Setup audio address and port */
@ -8419,6 +8380,26 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
ast_rtp_set_peer(p->rtp, &sin); ast_rtp_set_peer(p->rtp, &sin);
if (debug) if (debug)
ast_verbose("Peer audio RTP is at port %s:%d\n", ast_inet_ntoa(sin.sin_addr), ntohs(sin.sin_port)); ast_verbose("Peer audio RTP is at port %s:%d\n", ast_inet_ntoa(sin.sin_addr), ntohs(sin.sin_port));
/* 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->jointnoncodeccapability = newnoncodeccapability; /* DTMF capabilities */
ast_rtp_pt_copy(p->rtp, newaudiortp);
if (ast_test_flag(&p->flags[0], SIP_DTMF) == SIP_DTMF_AUTO) {
ast_clear_flag(&p->flags[0], SIP_DTMF);
if (newnoncodeccapability & AST_RTP_DTMF) {
/* XXX Would it be reasonable to drop the DSP at this point? XXX */
ast_set_flag(&p->flags[0], SIP_DTMF_RFC2833);
/* Since RFC2833 is now negotiated we need to change some properties of the RTP stream */
ast_rtp_setdtmf(p->rtp, 1);
ast_rtp_setdtmfcompensate(p->rtp, ast_test_flag(&p->flags[1], SIP_PAGE2_RFC2833_COMPENSATE));
} else {
ast_set_flag(&p->flags[0], SIP_DTMF_INBAND);
}
}
} else if (udptlportno > 0) { } else if (udptlportno > 0) {
if (debug) if (debug)
ast_verbose("Got T.38 Re-invite without audio. Keeping RTP active during T.38 session.\n"); ast_verbose("Got T.38 Re-invite without audio. Keeping RTP active during T.38 session.\n");
@ -8438,6 +8419,7 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
ast_rtp_set_peer(p->vrtp, &vsin); ast_rtp_set_peer(p->vrtp, &vsin);
if (debug) if (debug)
ast_verbose("Peer video RTP is at port %s:%d\n", ast_inet_ntoa(vsin.sin_addr), ntohs(vsin.sin_port)); ast_verbose("Peer video RTP is at port %s:%d\n", ast_inet_ntoa(vsin.sin_addr), ntohs(vsin.sin_port));
ast_rtp_pt_copy(p->vrtp, newvideortp);
} else { } else {
ast_rtp_stop(p->vrtp); ast_rtp_stop(p->vrtp);
if (debug) if (debug)
@ -8454,6 +8436,13 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
ast_rtp_set_peer(p->trtp, &tsin); ast_rtp_set_peer(p->trtp, &tsin);
if (debug) if (debug)
ast_verbose("Peer T.140 RTP is at port %s:%d\n", ast_inet_ntoa(vsin.sin_addr), ntohs(vsin.sin_port)); ast_verbose("Peer T.140 RTP is at port %s:%d\n", ast_inet_ntoa(vsin.sin_addr), ntohs(vsin.sin_port));
if ((p->jointcapability & AST_FORMAT_T140RED)) {
p->red = 1;
rtp_red_init(p->trtp, 300, red_data_pt, 2);
} else {
p->red = 0;
}
ast_rtp_pt_copy(p->trtp, newtextrtp);
} else { } else {
ast_rtp_stop(p->trtp); ast_rtp_stop(p->trtp);
if (debug) if (debug)
@ -8474,10 +8463,21 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
ast_log(LOG_DEBUG, "Peer T.38 UDPTL is set behind NAT and with destination, destination address now %s\n", ast_inet_ntoa(isin.sin_addr)); ast_log(LOG_DEBUG, "Peer T.38 UDPTL is set behind NAT and with destination, destination address now %s\n", ast_inet_ntoa(isin.sin_addr));
} }
} }
} else {
memcpy(&isin.sin_addr, ihp->h_addr, sizeof(sin.sin_addr));
} }
ast_udptl_set_peer(p->udptl, &isin); ast_udptl_set_peer(p->udptl, &isin);
if (debug) if (debug)
ast_debug(1,"Peer T.38 UDPTL is at port %s:%d\n", ast_inet_ntoa(isin.sin_addr), ntohs(isin.sin_port)); ast_debug(1,"Peer T.38 UDPTL is at port %s:%d\n", ast_inet_ntoa(isin.sin_addr), ntohs(isin.sin_port));
/* Remote party offers T38, we need to update state */
if ((t38action == SDP_T38_ACCEPT) &&
(p->t38.state == T38_LOCAL_REINVITE)) {
change_t38_state(p, T38_ENABLED);
} else if ((t38action == SDP_T38_INITIATE) &&
p->owner && p->lastinvite) {
change_t38_state(p, T38_PEER_REINVITE); /* T38 Offered in re-invite from remote party */
}
} else { } else {
ast_udptl_stop(p->udptl); ast_udptl_stop(p->udptl);
if (debug) if (debug)
@ -8485,6 +8485,11 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
} }
} }
if ((portno == -1) && (p->t38.state != T38_DISABLED)) {
ast_debug(3, "Have T.38 but no audio, accepting offer anyway\n");
return 0;
}
/* Ok, we're going with this offer */ /* Ok, we're going with this offer */
ast_debug(2, "We're settling with these formats: %s\n", ast_getformatname_multiple(buf, SIPBUFSIZE, p->jointcapability)); ast_debug(2, "We're settling with these formats: %s\n", ast_getformatname_multiple(buf, SIPBUFSIZE, p->jointcapability));

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