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Fix T.38 negotiation regression introduced with the SDP parser changes. 

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@229912 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Joshua Colp 2009-11-13 15:56:16 +00:00
parent ebc3aff1c3
commit b3b6537e71
1 changed files with 60 additions and 57 deletions
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117
channels/chan_sip.c
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@ -8374,9 +8374,11 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
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 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
hp = &sessionhp.hp;
vhp = hp;
thp = hp;
ihp = hp;
}
break;
case 'a':
@ -8577,15 +8580,6 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
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);
@ -8619,6 +8613,11 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
processed = TRUE;
thp = &texthp.hp;
}
} else if (image) {
if (process_sdp_c(value, &imagehp)) {
processed = TRUE;
ihp = &imagehp.hp;
}
}
break;
case 'a':
@ -8708,54 +8707,10 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
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 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
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 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
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 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
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)
@ -8822,10 +8809,21 @@ 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));
}
}
} 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)
@ -8833,6 +8831,11 @@ 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 */
ast_debug(2, "We're settling with these formats: %s\n", ast_getformatname_multiple(buf, SIPBUFSIZE, p->jointcapability));