diff options
author | file <file@f38db490-d61c-443f-a65b-d21fe96a405b> | 2009-11-13 15:56:48 +0000 |
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committer | file <file@f38db490-d61c-443f-a65b-d21fe96a405b> | 2009-11-13 15:56:48 +0000 |
commit | 071893ff6e78b62c44d80362a93f562d81c06ebe (patch) | |
tree | e254d91dfdefabbbf3695f8c64ba5916213bfd96 | |
parent | 5cd409b4904162895d1da1015dad02a60fd822f5 (diff) |
Merged revisions 229912 via svnmerge from
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: http://svn.digium.com/svn/asterisk/branches/1.6.0@229913 f38db490-d61c-443f-a65b-d21fe96a405b
-rw-r--r-- | channels/chan_sip.c | 93 |
1 files changed, 50 insertions, 43 deletions
diff --git a/channels/chan_sip.c b/channels/chan_sip.c index 82039cd6c..c9dd6b71d 100644 --- a/channels/chan_sip.c +++ b/channels/chan_sip.c @@ -6891,9 +6891,11 @@ 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 */ @@ -6999,6 +7001,7 @@ 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': @@ -7113,15 +7116,6 @@ 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); @@ -7155,6 +7149,11 @@ 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': @@ -7239,41 +7238,11 @@ 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(s3, SIPBUFSIZE, 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 */ - - 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); - } + 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 */ @@ -7285,6 +7254,26 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action ast_rtp_set_peer(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 */ + + 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) { if (debug) ast_verbose("Got T.38 Re-invite without audio. Keeping RTP active during T.38 session.\n"); @@ -7304,6 +7293,7 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action ast_rtp_set_peer(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_pt_copy(p->vrtp, newvideortp); } else { ast_rtp_stop(p->vrtp); if (debug) @@ -7320,6 +7310,7 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action ast_rtp_set_peer(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)); + ast_rtp_pt_copy(p->trtp, newtextrtp); } else { ast_rtp_stop(p->trtp); if (debug) @@ -7340,10 +7331,21 @@ 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) @@ -7351,6 +7353,11 @@ 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)); |