[asterisk-commits] file: trunk r229912 - /trunk/channels/chan_sip.c
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Fri Nov 13 09:56:20 CST 2009
Author: file
Date: Fri Nov 13 09:56:16 2009
New Revision: 229912
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=229912
Log:
Fix T.38 negotiation regression introduced with the SDP parser changes.
Modified:
trunk/channels/chan_sip.c
Modified: trunk/channels/chan_sip.c
URL: http://svnview.digium.com/svn/asterisk/trunk/channels/chan_sip.c?view=diff&rev=229912&r1=229911&r2=229912
==============================================================================
--- trunk/channels/chan_sip.c (original)
+++ trunk/channels/chan_sip.c Fri Nov 13 09:56:16 2009
@@ -8374,9 +8374,11 @@
struct ast_hostent audiohp;
struct ast_hostent videohp;
struct ast_hostent texthp;
+ struct ast_hostent imagehp;
struct hostent *hp = NULL; /*!< RTP Audio host IP */
struct hostent *vhp = NULL; /*!< RTP video 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 vportno = -1; /*!< RTP Video port number */
int tportno = -1; /*!< RTP Text port number */
@@ -8463,6 +8465,7 @@
hp = &sessionhp.hp;
vhp = hp;
thp = hp;
+ ihp = hp;
}
break;
case 'a':
@@ -8577,15 +8580,6 @@
if (p->t38.state != T38_ENABLED) {
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 {
ast_log(LOG_WARNING, "Unsupported SDP media type in offer: %s\n", m);
@@ -8618,6 +8612,11 @@
if (process_sdp_c(value, &texthp)) {
processed = TRUE;
thp = &texthp.hp;
+ }
+ } else if (image) {
+ if (process_sdp_c(value, &imagehp)) {
+ processed = TRUE;
+ ihp = &imagehp.hp;
}
}
break;
@@ -8708,54 +8707,10 @@
ast_rtp_lookup_mime_multiple2(s2, peernoncodeccapability, 0, 0),
ast_rtp_lookup_mime_multiple2(s3, newnoncodeccapability, 0, 0));
}
- if (!newjointcapability) {
- /* If T.38 was not negotiated either, totally bail out... */
- if ((p->t38.state == T38_DISABLED) || !udptlportno) {
- ast_log(LOG_NOTICE, "No compatible codecs, not accepting this offer!\n");
- /* 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 (ast_test_flag(&p->flags[1], SIP_PAGE2_PREFERRED_CODEC)) { /* respond with single most preferred joint codec, limiting the other side's choice */
- p->jointcapability = ast_codec_choose(&p->prefs, p->jointcapability, 1);
- }
-
- if (p->trtp && (p->jointcapability & AST_FORMAT_T140RED)) {
- p->red = 1;
- ast_rtp_red_init(p->trtp, 300, red_data_pt, 2);
- } else {
- p->red = 0;
- }
-
- ast_rtp_codecs_payloads_copy(&newaudiortp, ast_rtp_instance_get_codecs(p->rtp), p->rtp);
- if (p->vrtp) {
- ast_rtp_codecs_payloads_copy(&newvideortp, ast_rtp_instance_get_codecs(p->vrtp), p->vrtp);
- }
- if (p->trtp) {
- ast_rtp_codecs_payloads_copy(&newtextrtp, ast_rtp_instance_get_codecs(p->trtp), p->trtp);
- }
-
- 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_instance_set_prop(p->rtp, AST_RTP_PROPERTY_DTMF, 1);
- ast_rtp_instance_set_prop(p->rtp, AST_RTP_PROPERTY_DTMF_COMPENSATE, ast_test_flag(&p->flags[1], SIP_PAGE2_RFC2833_COMPENSATE));
- } else {
- ast_set_flag(&p->flags[0], SIP_DTMF_INBAND);
- }
+ if (!newjointcapability && (portno != -1)) {
+ ast_log(LOG_NOTICE, "No compatible codecs, not accepting this offer!\n");
+ /* Do NOT Change current setting */
+ return -1;
}
/* Setup audio address and port */
@@ -8767,6 +8722,30 @@
ast_rtp_instance_set_remote_address(p->rtp, &sin);
if (debug)
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 */
+
+ if (ast_test_flag(&p->flags[1], SIP_PAGE2_PREFERRED_CODEC)) { /* respond with single most preferred joint codec, limiting the other side's choice */
+ p->jointcapability = ast_codec_choose(&p->prefs, p->jointcapability, 1);
+ }
+
+ ast_rtp_codecs_payloads_copy(&newaudiortp, ast_rtp_instance_get_codecs(p->rtp), p->rtp);
+
+ 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_instance_set_prop(p->rtp, AST_RTP_PROPERTY_DTMF, 1);
+ ast_rtp_instance_set_prop(p->rtp, AST_RTP_PROPERTY_DTMF_COMPENSATE, ast_test_flag(&p->flags[1], SIP_PAGE2_RFC2833_COMPENSATE));
+ } else {
+ ast_set_flag(&p->flags[0], SIP_DTMF_INBAND);
+ }
+ }
} else if (udptlportno > 0) {
if (debug)
ast_verbose("Got T.38 Re-invite without audio. Keeping RTP active during T.38 session.\n");
@@ -8786,6 +8765,7 @@
ast_rtp_instance_set_remote_address(p->vrtp, &vsin);
if (debug)
ast_verbose("Peer video RTP is at port %s:%d\n", ast_inet_ntoa(vsin.sin_addr), ntohs(vsin.sin_port));
+ ast_rtp_codecs_payloads_copy(&newvideortp, ast_rtp_instance_get_codecs(p->vrtp), p->vrtp);
} else {
ast_rtp_instance_stop(p->vrtp);
if (debug)
@@ -8802,6 +8782,13 @@
ast_rtp_instance_set_remote_address(p->trtp, &tsin);
if (debug)
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;
+ ast_rtp_red_init(p->trtp, 300, red_data_pt, 2);
+ } else {
+ p->red = 0;
+ }
+ ast_rtp_codecs_payloads_copy(&newtextrtp, ast_rtp_instance_get_codecs(p->trtp), p->trtp);
} else {
ast_rtp_instance_stop(p->trtp);
if (debug)
@@ -8822,16 +8809,32 @@
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);
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));
+
+ /* 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 {
ast_udptl_stop(p->udptl);
if (debug)
ast_debug(1, "Peer doesn't provide T.38 UDPTL\n");
}
}
+
+ 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 */
ast_debug(2, "We're settling with these formats: %s\n", ast_getformatname_multiple(buf, SIPBUFSIZE, p->jointcapability));
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