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Merge of the "sdpcleanup" branch. Thanks to John Martin for a lot of tests

Browse Source 
and some patches (all disclaimed).

- Don't change RTP properties if we reject a re-INVITE
- Don't add video to an outbound channel if there's no video on the inbound channel
- Don't include video in the "preferred codec" list for codec selection
- Clean up and document code that parses and adds SDP attachments

Since we do not transcode video, we can't handle video the same way as audio. This is a
bug fix patch. In future releases, we need to work on a solution for video negotiation,
not codecs but formats and framerates instead.



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@32597 65c4cc65-6c06-0410-ace0-fbb531ad65f3
1.4
Olle Johansson 20 years ago
parent 016034115e
commit 479181951a
6 changed files with 459 additions and 237 deletions
Show Stats Download Patch File Download Diff File

9
channel.c
Unescape View File

@ -539,7 +539,7 @@ char *ast_transfercapability2str(int transfercapability)
} }
} }
/*! \brief Pick the best codec */ /*! \brief Pick the best audio codec */
int ast_best_codec(int fmts) int ast_best_codec(int fmts)
{ {
/* This just our opinion, expressed in code. We are asked to choose /* This just our opinion, expressed in code. We are asked to choose
@ -573,6 +573,8 @@ int ast_best_codec(int fmts)
AST_FORMAT_G723_1, AST_FORMAT_G723_1,
}; };
/* Strip out video */
fmts &= AST_FORMAT_AUDIO_MASK;
/* Find the first preferred codec in the format given */ /* Find the first preferred codec in the format given */
for (x=0; x < (sizeof(prefs) / sizeof(prefs[0]) ); x++) for (x=0; x < (sizeof(prefs) / sizeof(prefs[0]) ); x++)
@ -2614,6 +2616,7 @@ struct ast_channel *ast_request(const char *type, int format, void *data, int *c
int fmt; int fmt;
int res; int res;
int foo; int foo;
int videoformat = format & AST_FORMAT_VIDEO_MASK;
if (!cause) if (!cause)
cause = &foo; cause = &foo;
@ -2629,7 +2632,7 @@ struct ast_channel *ast_request(const char *type, int format, void *data, int *c
continue; continue;
capabilities = chan->tech->capabilities; capabilities = chan->tech->capabilities;
fmt = format; fmt = format & AST_FORMAT_AUDIO_MASK;
res = ast_translator_best_choice(&fmt, &capabilities); res = ast_translator_best_choice(&fmt, &capabilities);
if (res < 0) { if (res < 0) {
ast_log(LOG_WARNING, "No translator path exists for channel type %s (native %d) to %d\n", type, chan->tech->capabilities, format); ast_log(LOG_WARNING, "No translator path exists for channel type %s (native %d) to %d\n", type, chan->tech->capabilities, format);
@ -2640,7 +2643,7 @@ struct ast_channel *ast_request(const char *type, int format, void *data, int *c
if (!chan->tech->requester) if (!chan->tech->requester)
return NULL; return NULL;
if (!(c = chan->tech->requester(type, capabilities, data, cause))) if (!(c = chan->tech->requester(type, capabilities | videoformat, data, cause)))
return NULL; return NULL;
if (c->_state == AST_STATE_DOWN) { if (c->_state == AST_STATE_DOWN) {

611
channels/chan_sip.c
Unescape View File

@ -1260,6 +1260,7 @@ static struct ast_rtp *sip_get_vrtp_peer(struct ast_channel *chan);
static int sip_get_codec(struct ast_channel *chan); static int sip_get_codec(struct ast_channel *chan);
static struct ast_frame *sip_rtp_read(struct ast_channel *ast, struct sip_pvt *p); static struct ast_frame *sip_rtp_read(struct ast_channel *ast, struct sip_pvt *p);
/*! \brief Definition of this channel for PBX channel registration */ /*! \brief Definition of this channel for PBX channel registration */
static const struct ast_channel_tech sip_tech = { static const struct ast_channel_tech sip_tech = {
.type = "SIP", .type = "SIP",
@ -3252,13 +3253,17 @@ static int sip_indicate(struct ast_channel *ast, int condition, const void *data
/*! \brief Initiate a call in the SIP channel /*! \brief Initiate a call in the SIP channel
called from sip_request_call (calls from the pbx ) */ called from sip_request_call (calls from the pbx ) for outbound channels
and from handle_request_invite for inbound channels
*/
static struct ast_channel *sip_new(struct sip_pvt *i, int state, const char *title) static struct ast_channel *sip_new(struct sip_pvt *i, int state, const char *title)
{ {
struct ast_channel *tmp; struct ast_channel *tmp;
struct ast_variable *v = NULL; struct ast_variable *v = NULL;
int fmt; int fmt;
int what; int what;
int needvideo = 0;
ast_mutex_unlock(&i->lock); ast_mutex_unlock(&i->lock);
/* Don't hold a sip pvt lock while we allocate a channel */ /* Don't hold a sip pvt lock while we allocate a channel */
@ -3269,17 +3274,50 @@ static struct ast_channel *sip_new(struct sip_pvt *i, int state, const char *tit
return NULL; return NULL;
} }
tmp->tech = &sip_tech; tmp->tech = &sip_tech;
/* Select our native format based on codec preference until we receive /* Select our native format based on codec preference until we receive
something from another device to the contrary. */ something from another device to the contrary. */
if (i->jointcapability) if (i->jointcapability) /* The joint capabilities of us and peer */
what = i->jointcapability; what = i->jointcapability;
else if (i->capability) else if (i->capability) /* Our configured capability for this peer */
what = i->capability; what = i->capability;
else else
what = global_capability; what = global_capability; /* Global codec support */
/* Set the native formats for audio and merge in video */
tmp->nativeformats = ast_codec_choose(&i->prefs, what, 1) | (i->jointcapability & AST_FORMAT_VIDEO_MASK); tmp->nativeformats = ast_codec_choose(&i->prefs, what, 1) | (i->jointcapability & AST_FORMAT_VIDEO_MASK);
if (option_debug > 2) {
char buf[BUFSIZ];
ast_log(LOG_DEBUG, "*** Our native formats are %s \n", ast_getformatname_multiple(buf, BUFSIZ, tmp->nativeformats));
ast_log(LOG_DEBUG, "*** Joint capabilities are %s \n", ast_getformatname_multiple(buf, BUFSIZ, i->jointcapability));
ast_log(LOG_DEBUG, "*** Our capabilities are %s \n", ast_getformatname_multiple(buf, BUFSIZ, i->capability));
ast_log(LOG_DEBUG, "*** AST_CODEC_CHOOSE formats are %s \n", ast_getformatname_multiple(buf, BUFSIZ, ast_codec_choose(&i->prefs, what, 1)));
if (i->prefcodec)
ast_log(LOG_DEBUG, "*** Our preferred formats from the incoming channel are %s \n", ast_getformatname_multiple(buf, BUFSIZ, i->prefcodec));
}
/* XXX Why are we choosing a codec from the native formats?? */
fmt = ast_best_codec(tmp->nativeformats); fmt = ast_best_codec(tmp->nativeformats);
/* If we have a prefcodec setting, we have an inbound channel that set a
preferred format for this call. Otherwise, we check the jointcapability
We also check for vrtp. If it's not there, we are not allowed do any video anyway.
*/
if (i->vrtp) {
if (i->prefcodec)
needvideo = i->prefcodec & AST_FORMAT_VIDEO_MASK; /* Outbound call */
else
needvideo = i->jointcapability & AST_FORMAT_VIDEO_MASK; /* Inbound call */
}
if (option_debug > 2) {
if (needvideo)
ast_log(LOG_DEBUG, "This channel can handle video! HOLLYWOOD next!\n");
else
ast_log(LOG_DEBUG, "This channel will not be able to handle video.\n");
}
if (title) if (title)
ast_string_field_build(tmp, name, "SIP/%s-%04lx", title, ast_random() & 0xffff); ast_string_field_build(tmp, name, "SIP/%s-%04lx", title, ast_random() & 0xffff);
else if (strchr(i->fromdomain,':')) else if (strchr(i->fromdomain,':'))
@ -3297,7 +3335,7 @@ static struct ast_channel *sip_new(struct sip_pvt *i, int state, const char *tit
tmp->fds[0] = ast_rtp_fd(i->rtp); tmp->fds[0] = ast_rtp_fd(i->rtp);
tmp->fds[1] = ast_rtcp_fd(i->rtp); tmp->fds[1] = ast_rtcp_fd(i->rtp);
} }
if (i->vrtp) { if (needvideo && i->vrtp) {
tmp->fds[2] = ast_rtp_fd(i->vrtp); tmp->fds[2] = ast_rtp_fd(i->vrtp);
tmp->fds[3] = ast_rtcp_fd(i->vrtp); tmp->fds[3] = ast_rtcp_fd(i->vrtp);
} }
@ -4038,7 +4076,9 @@ static int find_sdp(struct sip_request *req)
return 0; return 0;
} }
/*! \brief Process SIP SDP, select formats and activate RTP channels */ /*! \brief Process SIP SDP offer, select formats and activate RTP channels
If offer is rejected, we will not change any properties of the call
*/
static int process_sdp(struct sip_pvt *p, struct sip_request *req) static int process_sdp(struct sip_pvt *p, struct sip_request *req)
{ {
const char *m; /* SDP media offer */ const char *m; /* SDP media offer */
@ -4047,33 +4087,49 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req)
char host[258]; char host[258];
char iabuf[INET_ADDRSTRLEN]; char iabuf[INET_ADDRSTRLEN];
int len = -1; int len = -1;
int portno = -1; /* Audio port */ int portno = -1; /*!< RTP Audio port number */
int vportno = -1; /* Video port */ int vportno = -1; /*!< RTP Video port number */
/* Peer capability is the capability in the SDP, non codec is RFC2833 DTMF (101) */ /* Peer capability is the capability in the SDP, non codec is RFC2833 DTMF (101) */
int peercapability, peernoncodeccapability; int peercapability, peernoncodeccapability;
int vpeercapability = 0, vpeernoncodeccapability = 0;
struct sockaddr_in sin; /*!< media socket address */
struct sockaddr_in vsin; /*!< Video socket address */
int vpeercapability=0, vpeernoncodeccapability=0; /* Peer's video capabilities */
struct sockaddr_in sin;
const char *codecs; const char *codecs;
struct hostent *hp; struct hostent *hp; /*!< RTP Audio host IP */
struct ast_hostent ahp; struct hostent *vhp = NULL; /*!< RTP video host IP */
struct ast_hostent audiohp;
struct ast_hostent videohp;
int codec; int codec;
int destiterator = 0; int destiterator = 0;
int iterator; int iterator;
int sendonly = 0; int sendonly = 0;
int x, y; int numberofports;
int debug = sip_debug_test_pvt(p);
struct ast_channel *bridgepeer = NULL; struct ast_channel *bridgepeer = NULL;
struct ast_rtp newaudiortp, newvideortp; /* Buffers for codec handling */
int newjointcapability; /* Negotiated capability */
int newpeercapability;
int newnoncodeccapability;
int numberofmediastreams = 0;
int debug = sip_debug_test_pvt(p);
if (!p->rtp) { if (!p->rtp) {
ast_log(LOG_ERROR, "Got SDP but have no RTP session allocated.\n"); ast_log(LOG_ERROR, "Got SDP but have no RTP session allocated.\n");
return -1; return -1;
} }
/* Initialize the temporary RTP structures we use to evaluate the offer from the peer */
memset(&newaudiortp, 0, sizeof(newaudiortp));
memset(&newvideortp, 0, sizeof(newvideortp));
ast_rtp_pt_default(&newaudiortp);
ast_rtp_pt_default(&newvideortp);
/* Update our last rtprx when we receive an SDP, too */ /* Update our last rtprx when we receive an SDP, too */
p->lastrtprx = p->lastrtptx = time(NULL); /* XXX why both ? */ p->lastrtprx = p->lastrtptx = time(NULL); /* XXX why both ? */
/* Try to find first media stream */
m = get_sdp(req, "m"); m = get_sdp(req, "m");
destiterator = req->sdp_start; destiterator = req->sdp_start;
c = get_sdp_iterate(&destiterator, req, "c"); c = get_sdp_iterate(&destiterator, req, "c");
@ -4081,26 +4137,41 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req)
ast_log(LOG_WARNING, "Insufficient information for SDP (m = '%s', c = '%s')\n", m, c); ast_log(LOG_WARNING, "Insufficient information for SDP (m = '%s', c = '%s')\n", m, c);
return -1; return -1;
} }
/* Check for IPv4 address (not IPv6 yet) */
if (sscanf(c, "IN IP4 %256s", host) != 1) { if (sscanf(c, "IN IP4 %256s", host) != 1) {
ast_log(LOG_WARNING, "Invalid host in c= line, '%s'\n", c); ast_log(LOG_WARNING, "Invalid host in c= line, '%s'\n", c);
return -1; return -1;
} }
/* XXX This could block for a long time, and block the main thread! XXX */ /* XXX This could block for a long time, and block the main thread! XXX */
hp = ast_gethostbyname(host, &ahp); hp = ast_gethostbyname(host, &audiohp);
if (!hp) { if (!hp) {
ast_log(LOG_WARNING, "Unable to lookup host in c= line, '%s'\n", c); ast_log(LOG_WARNING, "Unable to lookup host in c= line, '%s'\n", c);
return -1; return -1;
} }
vhp = hp; /* Copy to video address as default too */
iterator = req->sdp_start; iterator = req->sdp_start;
ast_set_flag(&p->flags[0], SIP_NOVIDEO); ast_set_flag(&p->flags[0], SIP_NOVIDEO);
/* Find media streams in this SDP offer */
while ((m = get_sdp_iterate(&iterator, req, "m"))[0] != '\0') { while ((m = get_sdp_iterate(&iterator, req, "m"))[0] != '\0') {
int found = 0; int x;
if ((sscanf(m, "audio %d/%d RTP/AVP %n", &x, &y, &len) == 2) || int audio = FALSE;
numberofmediastreams++;
if (p->vrtp)
ast_rtp_pt_clear(&newvideortp); /* Must be cleared in case no m=video line exists */
numberofports = 1;
if ((sscanf(m, "audio %d/%d RTP/AVP %n", &x, &numberofports, &len) == 2) ||
(sscanf(m, "audio %d RTP/AVP %n", &x, &len) == 1)) { (sscanf(m, "audio %d RTP/AVP %n", &x, &len) == 1)) {
found = 1; audio = TRUE;
/* Found audio stream in this media definition */
portno = x; portno = x;
/* Scan through the RTP payload types specified in a "m=" line: */ /* Scan through the RTP payload types specified in a "m=" line: */
ast_rtp_pt_clear(p->rtp); ast_rtp_pt_clear(&newaudiortp);
for (codecs = m + len; !ast_strlen_zero(codecs); codecs = ast_skip_blanks(codecs + len)) { for (codecs = m + len; !ast_strlen_zero(codecs); codecs = ast_skip_blanks(codecs + len)) {
if (sscanf(codecs, "%d%n", &codec, &len) != 1) { if (sscanf(codecs, "%d%n", &codec, &len) != 1) {
ast_log(LOG_WARNING, "Error in codec string '%s'\n", codecs); ast_log(LOG_WARNING, "Error in codec string '%s'\n", codecs);
@ -4108,14 +4179,11 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req)
} }
if (debug) if (debug)
ast_verbose("Found RTP audio format %d\n", codec); ast_verbose("Found RTP audio format %d\n", codec);
ast_rtp_set_m_type(p->rtp, codec); ast_rtp_set_m_type(&newaudiortp, codec);
}
} }
if (p->vrtp) } else if ((sscanf(m, "video %d/%d RTP/AVP %n", &x, &numberofports, &len) == 2) ||
ast_rtp_pt_clear(p->vrtp); /* Must be cleared in case no m=video line exists */ (sscanf(m, "video %d RTP/AVP %n", &x, &len) == 1)) {
/* If it is not audio - is it video ? */
if (p->vrtp && (sscanf(m, "video %d RTP/AVP %n", &x, &len) == 1)) {
found = 1;
ast_clear_flag(&p->flags[0], SIP_NOVIDEO); ast_clear_flag(&p->flags[0], SIP_NOVIDEO);
vportno = x; vportno = x;
/* Scan through the RTP payload types specified in a "m=" line: */ /* Scan through the RTP payload types specified in a "m=" line: */
@ -4126,102 +4194,166 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req)
} }
if (debug) if (debug)
ast_verbose("Found RTP video format %d\n", codec); ast_verbose("Found RTP video format %d\n", codec);
ast_rtp_set_m_type(p->vrtp, codec); ast_rtp_set_m_type(&newvideortp, codec);
}
}
if (!found )
ast_log(LOG_WARNING, "Unknown SDP media type in offer: %s\n", m);
}
if (portno == -1 && vportno == -1) {
/* No acceptable offer found in SDP */
return -2;
} }
} else
ast_log(LOG_WARNING, "Unsupported SDP media type in offer: %s\n", m);
if (numberofports > 1)
ast_log(LOG_WARNING, "SDP offered %d ports for media, not supported by Asterisk. Will try anyway...\n", numberofports);
/* Check for Media-description-level-address for audio */ /* Check for Media-description-level-address for audio */
if (pedanticsipchecking) {
c = get_sdp_iterate(&destiterator, req, "c"); c = get_sdp_iterate(&destiterator, req, "c");
if (!ast_strlen_zero(c)) { if (!ast_strlen_zero(c)) {
if (sscanf(c, "IN IP4 %256s", host) != 1) { if (sscanf(c, "IN IP4 %256s", host) != 1) {
ast_log(LOG_WARNING, "Invalid secondary host in c= line, '%s'\n", c); ast_log(LOG_WARNING, "Invalid secondary host in c= line, '%s'\n", c);
} else { } else {
/* XXX This could block for a long time, and block the main thread! XXX */ /* XXX This could block for a long time, and block the main thread! XXX */
hp = ast_gethostbyname(host, &ahp); if (audio) {
if (!hp) { if ( !(hp = ast_gethostbyname(host, &audiohp)))
ast_log(LOG_WARNING, "Unable to lookup host in secondary c= line, '%s'\n", c); ast_log(LOG_WARNING, "Unable to lookup RTP Audio host in secondary c= line, '%s'\n", c);
} } else if (!(vhp = ast_gethostbyname(host, &videohp)))
ast_log(LOG_WARNING, "Unable to lookup RTP video host in secondary c= line, '%s'\n", c);
} }
} }
} }
if (portno == -1 && vportno == -1)
/* No acceptable offer found in SDP - we have no ports */
/* Do not change RTP or VRTP if this is a re-invite */
return -2;
if (numberofmediastreams > 2)
/* We have too many media streams, fail this offer */
return -3;
/* RTP addresses and ports for audio and video */ /* RTP addresses and ports for audio and video */
sin.sin_family = AF_INET; sin.sin_family = AF_INET;
vsin.sin_family = AF_INET;
memcpy(&sin.sin_addr, hp->h_addr, sizeof(sin.sin_addr)); memcpy(&sin.sin_addr, hp->h_addr, sizeof(sin.sin_addr));
if (vhp)
memcpy(&vsin.sin_addr, vhp->h_addr, sizeof(vsin.sin_addr));
/* Setup audio port number */ /* Setup audio port number */
sin.sin_port = htons(portno); sin.sin_port = htons(portno);
if (p->rtp && sin.sin_port) {
ast_rtp_set_peer(p->rtp, &sin);
if (debug) {
ast_verbose("Peer audio RTP is at port %s:%d\n", ast_inet_ntoa(iabuf,sizeof(iabuf), sin.sin_addr), ntohs(sin.sin_port));
}
}
/* Check for Media-description-level-address for video */
if (pedanticsipchecking) {
c = get_sdp_iterate(&destiterator, req, "c");
if (!ast_strlen_zero(c)) {
if (sscanf(c, "IN IP4 %256s", host) != 1) {
ast_log(LOG_WARNING, "Invalid secondary host in c= line, '%s'\n", c);
} else {
/* XXX This could block for a long time, and block the main thread! XXX */
hp = ast_gethostbyname(host, &ahp);
if (!hp) {
ast_log(LOG_WARNING, "Unable to lookup host in secondary c= line, '%s'\n", c);
}
}
}
}
/* Setup video port number */ /* Setup video port number */
sin.sin_port = htons(vportno); if (vportno != -1)
if (p->vrtp && sin.sin_port) { vsin.sin_port = htons(vportno);
ast_rtp_set_peer(p->vrtp, &sin);
if (debug) {
ast_verbose("Peer video RTP is at port %s:%d\n", ast_inet_ntoa(iabuf,sizeof(iabuf), sin.sin_addr), ntohs(sin.sin_port));
}
}
/* Next, scan through each "a=rtpmap:" line, noting each /* Next, scan through each "a=rtpmap:" line, noting each
* specified RTP payload type (with corresponding MIME subtype): * specified RTP payload type (with corresponding MIME subtype):
*/ */
/* XXX This needs to be done per media stream, since it's media stream specific */
iterator = req->sdp_start; iterator = req->sdp_start;
while ((a = get_sdp_iterate(&iterator, req, "a"))[0] != '\0') { while ((a = get_sdp_iterate(&iterator, req, "a"))[0] != '\0') {
char* mimeSubtype = ast_strdupa(a); /* ensures we have enough space */ char* mimeSubtype = ast_strdupa(a); /* ensures we have enough space */
if (!strcasecmp(a, "sendonly")) { if (!strcasecmp(a, "sendonly")) {
sendonly = 1; sendonly = 1;
continue; continue;
} } else if (!strcasecmp(a, "sendrecv")) {
if (!strcasecmp(a, "sendrecv")) {
sendonly = 0; sendonly = 0;
continue;
} else if (option_debug) {
/* If we're debugging, check for unsupported sdp options */
if (!strcasecmp(a, "inactive")) {
/* Inactive media streams: Not supported */
if (debug)
ast_verbose("Got unsupported a:inactive in SDP offer \n");
continue;
} else if (!strncasecmp(a, "rtcp:", (size_t) 5)) {
if (debug)
ast_verbose("Got unsupported a:rtcp in SDP offer \n");
continue;
} else if (!strncasecmp(a, "fmtp:", (size_t) 5)) {
/* Format parameters: Not supported */
/* Note: This is used for codec parameters, like bitrate for
G722 and video formats for H263 and H264
See RFC2327 for an example */
if (debug)
ast_verbose("Got unsupported a:fmtp in SDP offer \n");
continue;
} else if (!strncasecmp(a, "framerate:", (size_t) 10)) {
/* Video stuff: Not supported */
if (debug)
ast_verbose("Got unsupported a:framerate in SDP offer \n");
continue;
} else if (!strncasecmp(a, "maxprate:", (size_t) 9)) {
/* Video stuff: Not supported */
if (debug)
ast_verbose("Got unsupported a:maxprate in SDP offer \n");
continue;
} else if (!strncasecmp(a, "crypto:", (size_t) 7)) {
/* SRTP stuff, not yet supported */
if (debug)
ast_verbose("Got unsupported a:crypto in SDP offer \n");
continue;
} else if (!strncasecmp(a, "ptime:", (size_t) 6)) {
if (debug)
ast_verbose("Got unsupported a:ptime in SDP offer \n");
continue;
} }
if (sscanf(a, "rtpmap: %u %[^/]/", &codec, mimeSubtype) != 2) continue; } else if (sscanf(a, "rtpmap: %u %[^/]/", &codec, mimeSubtype) != 2)
continue;
/* We have a rtpmap to handle */
if (debug) if (debug)
ast_verbose("Found description format %s\n", mimeSubtype); ast_verbose("Found description format %s for ID %d\n", mimeSubtype, codec);
/* Note: should really look at the 'freq' and '#chans' params too */ /* Note: should really look at the 'freq' and '#chans' params too */
ast_rtp_set_rtpmap_type(p->rtp, codec, "audio", mimeSubtype); ast_rtp_set_rtpmap_type(&newaudiortp, codec, "audio", mimeSubtype);
if (p->vrtp) if (p->vrtp)
ast_rtp_set_rtpmap_type(p->vrtp, codec, "video", mimeSubtype); ast_rtp_set_rtpmap_type(&newvideortp, codec, "video", mimeSubtype);
} }
/* Now gather all of the codecs that were asked for: */ /* Now gather all of the codecs that we are asked for: */
ast_rtp_get_current_formats(p->rtp, ast_rtp_get_current_formats(&newaudiortp, &peercapability, &peernoncodeccapability);
&peercapability, &peernoncodeccapability); ast_rtp_get_current_formats(&newvideortp, &vpeercapability, &vpeernoncodeccapability);
newjointcapability = p->capability & (peercapability | vpeercapability);
newpeercapability = (peercapability | vpeercapability);
newnoncodeccapability = noncodeccapability & peernoncodeccapability;
if (debug) {
/* shame on whoever coded this.... */
char s1[BUFSIZ], s2[BUFSIZ], s3[BUFSIZ], s4[BUFSIZ];
ast_verbose("Capabilities: us - %s, peer - audio=%s/video=%s, combined - %s\n",
ast_getformatname_multiple(s1, BUFSIZ, p->capability),
ast_getformatname_multiple(s2, BUFSIZ, newpeercapability),
ast_getformatname_multiple(s3, BUFSIZ, vpeercapability),
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),
ast_rtp_lookup_mime_multiple(s2, BUFSIZ, peernoncodeccapability, 0),
ast_rtp_lookup_mime_multiple(s3, BUFSIZ, newnoncodeccapability, 0));
}
if (!newjointcapability) {
ast_log(LOG_NOTICE, "No compatible codecs, not accepting this offer!\n");
/* Do NOT Change current setting */
return -1;
}
/* 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 */
{
int i;
/* Copy payload types from source to destination */
for (i=0; i < MAX_RTP_PT; ++i) {
p->rtp->current_RTP_PT[i]= newaudiortp.current_RTP_PT[i];
if (p->vrtp) if (p->vrtp)
ast_rtp_get_current_formats(p->vrtp, p->vrtp->current_RTP_PT[i]= newvideortp.current_RTP_PT[i];
&vpeercapability, &vpeernoncodeccapability); }
p->jointcapability = p->capability & (peercapability | vpeercapability); }
p->peercapability = (peercapability | vpeercapability);
p->noncodeccapability = noncodeccapability & peernoncodeccapability;
if (ast_test_flag(&p->flags[0], SIP_DTMF) == SIP_DTMF_AUTO) { if (ast_test_flag(&p->flags[0], SIP_DTMF) == SIP_DTMF_AUTO) {
ast_clear_flag(&p->flags[0], SIP_DTMF); ast_clear_flag(&p->flags[0], SIP_DTMF);
if (p->noncodeccapability & AST_RTP_DTMF) { if (newnoncodeccapability & AST_RTP_DTMF) {
/* XXX Would it be reasonable to drop the DSP at this point? XXX */ /* XXX Would it be reasonable to drop the DSP at this point? XXX */
ast_set_flag(&p->flags[0], SIP_DTMF_RFC2833); ast_set_flag(&p->flags[0], SIP_DTMF_RFC2833);
} else { } else {
@ -4229,40 +4361,42 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req)
} }
} }
if (debug) { /* Setup audio port number */
/* shame on whoever coded this.... */ if (p->rtp && sin.sin_port) {
const unsigned slen=512; ast_rtp_set_peer(p->rtp, &sin);
char s1[slen], s2[slen], s3[slen], s4[slen]; if (debug)
ast_verbose("Peer audio RTP is at port %s:%d\n", ast_inet_ntoa(iabuf,sizeof(iabuf), sin.sin_addr), ntohs(sin.sin_port));
}
ast_verbose("Capabilities: us - %s, peer - audio=%s/video=%s, combined - %s\n", /* Setup video port number */
ast_getformatname_multiple(s1, slen, p->capability), if (p->vrtp && vsin.sin_port) {
ast_getformatname_multiple(s2, slen, peercapability), ast_rtp_set_peer(p->vrtp, &vsin);
ast_getformatname_multiple(s3, slen, vpeercapability), if (debug)
ast_getformatname_multiple(s4, slen, p->jointcapability)); ast_verbose("Peer video RTP is at port %s:%d\n", ast_inet_ntoa(iabuf,sizeof(iabuf), vsin.sin_addr), ntohs(vsin.sin_port));
ast_verbose("Non-codec capabilities: us - %s, peer - %s, combined - %s\n",
ast_rtp_lookup_mime_multiple(s1, slen, noncodeccapability, 0),
ast_rtp_lookup_mime_multiple(s2, slen, peernoncodeccapability, 0),
ast_rtp_lookup_mime_multiple(s3, slen, p->noncodeccapability, 0));
}
if (!p->jointcapability) {
ast_log(LOG_NOTICE, "No compatible codecs!\n");
return -1;
} }
if (!p->owner) /* There's no open channel owning us */ /* Ok, we're going with this offer */
if (option_debug > 1) {
char buf[BUFSIZ];
ast_log(LOG_DEBUG, "We're settling with these formats: %s\n", ast_getformatname_multiple(buf, BUFSIZ, p->jointcapability));
}
if (!p->owner) /* There's no open channel owning us so we can return here. For a re-invite or so, we proceed */
return 0; return 0;
if (!(p->owner->nativeformats & p->jointcapability & AST_FORMAT_AUDIO_MASK)) { if (!(p->owner->nativeformats & p->jointcapability & AST_FORMAT_AUDIO_MASK)) {
const unsigned slen=512; if (debug) {
char s1[slen], s2[slen]; char s1[BUFSIZ], s2[BUFSIZ];
ast_log(LOG_DEBUG, "Oooh, we need to change our formats since our peer supports only %s and not %s\n", ast_log(LOG_DEBUG, "Oooh, we need to change our audio formats since our peer supports only %s and not %s\n",
ast_getformatname_multiple(s1, slen, p->jointcapability), ast_getformatname_multiple(s1, BUFSIZ, p->jointcapability),
ast_getformatname_multiple(s2, slen, p->owner->nativeformats)); ast_getformatname_multiple(s2, BUFSIZ, p->owner->nativeformats));
}
p->owner->nativeformats = ast_codec_choose(&p->prefs, p->jointcapability, 1) | (p->capability & vpeercapability); p->owner->nativeformats = ast_codec_choose(&p->prefs, p->jointcapability, 1) | (p->capability & vpeercapability);
ast_set_read_format(p->owner, p->owner->readformat); ast_set_read_format(p->owner, p->owner->readformat);
ast_set_write_format(p->owner, p->owner->writeformat); ast_set_write_format(p->owner, p->owner->writeformat);
} }
if ((bridgepeer = ast_bridged_channel(p->owner))) { if ((bridgepeer = ast_bridged_channel(p->owner))) {
/* We have a bridge */ /* We have a bridge */
/* Turn on/off music on hold if we are holding/unholding */ /* Turn on/off music on hold if we are holding/unholding */
@ -4277,12 +4411,15 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req)
ast_moh_start(bridgepeer, NULL); ast_moh_start(bridgepeer, NULL);
if (sendonly) if (sendonly)
ast_rtp_stop(p->rtp); ast_rtp_stop(p->rtp);
/* RTCP needs to go ahead, even if we're on hold!!! */
/* Activate a re-invite */ /* Activate a re-invite */
ast_queue_frame(p->owner, &ast_null_frame); ast_queue_frame(p->owner, &ast_null_frame);
} }
} }
/* Manager Hold and Unhold events must be generated, if necessary */ /* Manager Hold and Unhold events must be generated, if necessary */
/* XXX Support for sendonly/recvonly needs to be fixed !!! */
if (sin.sin_addr.s_addr && !sendonly) { if (sin.sin_addr.s_addr && !sendonly) {
append_history(p, "Unhold", "%s", req->data); append_history(p, "Unhold", "%s", req->data);
@ -4309,9 +4446,11 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req)
ast_set_flag(&p->flags[0], SIP_CALL_ONHOLD); ast_set_flag(&p->flags[0], SIP_CALL_ONHOLD);
} }
return 0; return 0;
} }
/*! \brief Add header to SIP message */ /*! \brief Add header to SIP message */
static int add_header(struct sip_request *req, const char *var, const char *value) static int add_header(struct sip_request *req, const char *var, const char *value)
{ {
@ -4885,6 +5024,7 @@ static int add_vidupdate(struct sip_request *req)
return 0; return 0;
} }
/*! \brief Add codec offer to SDP offer/answer body in INVITE or 200 OK */
static void add_codec_to_sdp(const struct sip_pvt *p, int codec, int sample_rate, static void add_codec_to_sdp(const struct sip_pvt *p, int codec, int sample_rate,
char **m_buf, size_t *m_size, char **a_buf, size_t *a_size, char **m_buf, size_t *m_size, char **a_buf, size_t *a_size,
int debug) int debug)
@ -4905,6 +5045,7 @@ static void add_codec_to_sdp(const struct sip_pvt *p, int codec, int sample_rate
ast_build_string(a_buf, a_size, "a=fmtp:%d annexb=no\r\n", rtp_code); ast_build_string(a_buf, a_size, "a=fmtp:%d annexb=no\r\n", rtp_code);
} }
/*! \brief Add RFC 2833 DTMF offer to SDP */
static void add_noncodec_to_sdp(const struct sip_pvt *p, int format, int sample_rate, static void add_noncodec_to_sdp(const struct sip_pvt *p, int format, int sample_rate,
char **m_buf, size_t *m_size, char **a_buf, size_t *a_size, char **m_buf, size_t *m_size, char **a_buf, size_t *a_size,
int debug) int debug)
@ -4929,21 +5070,25 @@ static void add_noncodec_to_sdp(const struct sip_pvt *p, int format, int sample_
static int add_sdp(struct sip_request *resp, struct sip_pvt *p) static int add_sdp(struct sip_request *resp, struct sip_pvt *p)
{ {
int len = 0; int len = 0;
int pref_codec;
int alreadysent = 0; int alreadysent = 0;
struct sockaddr_in sin; struct sockaddr_in sin;
struct sockaddr_in vsin; struct sockaddr_in vsin;
char v[256]; struct sockaddr_in dest;
char s[256]; struct sockaddr_in vdest = { 0, };
char o[256];
char c[256]; /* SDP fields */
char t[256]; char *version = "v=0\r\n"; /* Protocol version */
char b[256]; char *subject = "s=session\r\n"; /* Subject of the session */
char owner[256]; /* Session owner/creator */
char connection[256]; /* Connection data */
char *stime = "t=0 0\r\n"; /* Time the session is active */
char bandwidth[256] = ""; /* Max bitrate */
char *hold; char *hold;
char m_audio[256]; char m_audio[256]; /* Media declaration line for audio */
char m_video[256]; char m_video[256]; /* Media declaration line for video */
char a_audio[1024]; char a_audio[1024]; /* Attributes for audio */
char a_video[1024]; char a_video[1024]; /* Attributes for video */
char *m_audio_next = m_audio; char *m_audio_next = m_audio;
char *m_video_next = m_video; char *m_video_next = m_video;
size_t m_audio_left = sizeof(m_audio); size_t m_audio_left = sizeof(m_audio);
@ -4952,31 +5097,33 @@ static int add_sdp(struct sip_request *resp, struct sip_pvt *p)
char *a_video_next = a_video; char *a_video_next = a_video;
size_t a_audio_left = sizeof(a_audio); size_t a_audio_left = sizeof(a_audio);
size_t a_video_left = sizeof(a_video); size_t a_video_left = sizeof(a_video);
char iabuf[INET_ADDRSTRLEN]; char iabuf[INET_ADDRSTRLEN];
int x; int x;
int capability; int capability;
struct sockaddr_in dest; int needvideo = FALSE;
struct sockaddr_in vdest = { 0, }; int debug = sip_debug_test_pvt(p);
int debug;
debug = sip_debug_test_pvt(p); m_video[0] = '\0'; /* Reset the video media string if it's not needed */
len = 0;
if (!p->rtp) { if (!p->rtp) {
ast_log(LOG_WARNING, "No way to add SDP without an RTP structure\n"); ast_log(LOG_WARNING, "No way to add SDP without an RTP structure\n");
return -1; return -1;
} }
capability = p->jointcapability;
/* Set RTP Session ID and version */
if (!p->sessionid) { if (!p->sessionid) {
p->sessionid = getpid(); p->sessionid = getpid();
p->sessionversion = p->sessionid; p->sessionversion = p->sessionid;
} else } else
p->sessionversion++; p->sessionversion++;
/* Get our addresses */
ast_rtp_get_us(p->rtp, &sin); ast_rtp_get_us(p->rtp, &sin);
if (p->vrtp) if (p->vrtp)
ast_rtp_get_us(p->vrtp, &vsin); ast_rtp_get_us(p->vrtp, &vsin);
/* Is this a re-invite to move the media out, then use the original offer from caller */
if (p->redirip.sin_addr.s_addr) { if (p->redirip.sin_addr.s_addr) {
dest.sin_port = p->redirip.sin_port; dest.sin_port = p->redirip.sin_port;
dest.sin_addr = p->redirip.sin_addr; dest.sin_addr = p->redirip.sin_addr;
@ -4987,60 +5134,109 @@ static int add_sdp(struct sip_request *resp, struct sip_pvt *p)
dest.sin_port = sin.sin_port; dest.sin_port = sin.sin_port;
} }
/* Determine video destination */ /* Ok, let's start working with codec selection here */
capability = p->jointcapability;
if (option_debug > 1) {
char codecbuf[BUFSIZ];
ast_log(LOG_DEBUG, "** Our capability: %s Video flag: %s\n", ast_getformatname_multiple(codecbuf, sizeof(codecbuf), capability), ast_test_flag(&p->flags[0], SIP_NOVIDEO) ? "True" : "False");
ast_log(LOG_DEBUG, "** Our prefcodec: %s \n", ast_getformatname_multiple(codecbuf, sizeof(codecbuf), p->prefcodec));
}
/* Check if we need video in this call */
if((capability & AST_FORMAT_VIDEO_MASK) && !ast_test_flag(&p->flags[0], SIP_NOVIDEO)) {
if (p->vrtp) { if (p->vrtp) {
needvideo = TRUE;
if (option_debug > 1)
ast_log(LOG_DEBUG, "This call needs video offers! \n");
} else if (option_debug > 1)
ast_log(LOG_DEBUG, "This call needs video offers, but there's no video support enabled ! \n");
}
/* Ok, we need video. Let's add what we need for video and set codecs.
Video is handled differently than audio since we can not transcode. */
if (needvideo) {
/* Determine video destination */
if (p->vredirip.sin_addr.s_addr) { if (p->vredirip.sin_addr.s_addr) {
vdest.sin_port = p->vredirip.sin_port;
vdest.sin_addr = p->vredirip.sin_addr; vdest.sin_addr = p->vredirip.sin_addr;
vdest.sin_port = p->vredirip.sin_port;
} else { } else {
vdest.sin_addr = p->ourip; vdest.sin_addr = p->ourip;
vdest.sin_port = vsin.sin_port; vdest.sin_port = vsin.sin_port;
} }
} ast_build_string(&m_video_next, &m_video_left, "m=video %d RTP/AVP", ntohs(vdest.sin_port));
if (debug) {
ast_verbose("We're at %s port %d\n", ast_inet_ntoa(iabuf, sizeof(iabuf), p->ourip), ntohs(sin.sin_port)); /* Build max bitrate string */
if (p->vrtp) if (p->maxcallbitrate)
snprintf(bandwidth, sizeof(bandwidth), "b=CT:%d\r\n", p->maxcallbitrate);
if (debug)
ast_verbose("Video is at %s port %d\n", ast_inet_ntoa(iabuf, sizeof(iabuf), p->ourip), ntohs(vsin.sin_port)); ast_verbose("Video is at %s port %d\n", ast_inet_ntoa(iabuf, sizeof(iabuf), p->ourip), ntohs(vsin.sin_port));
/* For video, we can't negotiate video offers. Let's compare the incoming call with what we got. */
if (p->prefcodec) {
int videocapability = (capability & p->prefcodec) & AST_FORMAT_VIDEO_MASK; /* Outbound call */
/*! \todo XXX We need to select one codec, not many, since there's no transcoding */
/* Now, merge this video capability into capability while removing unsupported codecs */
if (!videocapability) {
needvideo = FALSE;
if (option_debug > 2)
ast_log(LOG_DEBUG, "** No compatible video codecs... Disabling video.\n");
}
/* Replace video capabilities with the new videocapability */
capability = (capability & AST_FORMAT_AUDIO_MASK) | videocapability;
if (option_debug > 4) {
char codecbuf[BUFSIZ];
if (videocapability)
ast_log(LOG_DEBUG, "** Our video codec selection is: %s \n", ast_getformatname_multiple(codecbuf, sizeof(codecbuf), videocapability));
ast_log(LOG_DEBUG, "** Capability now set to : %s \n", ast_getformatname_multiple(codecbuf, sizeof(codecbuf), capability));
}
} }
}
if (debug)
ast_verbose("Audio is at %s port %d\n", ast_inet_ntoa(iabuf, sizeof(iabuf), p->ourip), ntohs(sin.sin_port));
/* Start building generic SDP headers */
/* We break with the "recommendation" and send our IP, in order that our /* We break with the "recommendation" and send our IP, in order that our
peer doesn't have to ast_gethostbyname() us */ peer doesn't have to ast_gethostbyname() us */
snprintf(v, sizeof(v), "v=0\r\n"); snprintf(owner, sizeof(owner), "o=root %d %d IN IP4 %s\r\n", p->sessionid, p->sessionversion, ast_inet_ntoa(iabuf, sizeof(iabuf), dest.sin_addr));
snprintf(o, sizeof(o), "o=root %d %d IN IP4 %s\r\n", p->sessionid, p->sessionversion, ast_inet_ntoa(iabuf, sizeof(iabuf), dest.sin_addr)); snprintf(connection, sizeof(connection), "c=IN IP4 %s\r\n", ast_inet_ntoa(iabuf, sizeof(iabuf), dest.sin_addr));
snprintf(s, sizeof(s), "s=session\r\n");
snprintf(c, sizeof(c), "c=IN IP4 %s\r\n", ast_inet_ntoa(iabuf, sizeof(iabuf), dest.sin_addr));
if ((p->vrtp) &&
(!ast_test_flag(&p->flags[0], SIP_NOVIDEO)) &&
(capability & VIDEO_CODEC_MASK)) /* only if video response is appropriate */
snprintf(b, sizeof(b), "b=CT:%d\r\n", p->maxcallbitrate);
snprintf(t, sizeof(t), "t=0 0\r\n");
ast_build_string(&m_audio_next, &m_audio_left, "m=audio %d RTP/AVP", ntohs(dest.sin_port)); ast_build_string(&m_audio_next, &m_audio_left, "m=audio %d RTP/AVP", ntohs(dest.sin_port));
ast_build_string(&m_video_next, &m_video_left, "m=video %d RTP/AVP", ntohs(vdest.sin_port));
if (ast_test_flag(&p->flags[0], SIP_CALL_ONHOLD)) if (ast_test_flag(&p->flags[0], SIP_CALL_ONHOLD))
hold = "a=recvonly\r\n"; hold = "a=recvonly\r\n";
else else
hold = "a=sendrecv\r\n"; hold = "a=sendrecv\r\n";
/* Prefer the codec we were requested to use, first, no matter what */ /* Now, start adding audio codecs. These are added in this order:
- First what was requested by the calling channel
- Then preferences in order from sip.conf device config for this peer/user
- Then other codecs in capabilities, including video
*/
/* Prefer the audio codec we were requested to use, first, no matter what
Note that p->prefcodec can include video codecs, so mask them out
*/
if (capability & p->prefcodec) { if (capability & p->prefcodec) {
if (p->prefcodec <= AST_FORMAT_MAX_AUDIO) add_codec_to_sdp(p, p->prefcodec & AST_FORMAT_AUDIO_MASK, 8000,
add_codec_to_sdp(p, p->prefcodec, 8000,
&m_audio_next, &m_audio_left, &m_audio_next, &m_audio_left,
&a_audio_next, &a_audio_left, &a_audio_next, &a_audio_left,
debug); debug);
else alreadysent |= p->prefcodec & AST_FORMAT_AUDIO_MASK;
add_codec_to_sdp(p, p->prefcodec, 90000,
&m_video_next, &m_video_left,
&a_video_next, &a_video_left,
debug);
alreadysent |= p->prefcodec;
} }
/* Start by sending our preferred codecs */
/* Start by sending our preferred audio codecs */
for (x = 0; x < 32; x++) { for (x = 0; x < 32; x++) {
int pref_codec;
if (!(pref_codec = ast_codec_pref_index(&p->prefs, x))) if (!(pref_codec = ast_codec_pref_index(&p->prefs, x)))
break; break;
@ -5050,27 +5246,21 @@ static int add_sdp(struct sip_request *resp, struct sip_pvt *p)
if (alreadysent & pref_codec) if (alreadysent & pref_codec)
continue; continue;
if (pref_codec <= AST_FORMAT_MAX_AUDIO)
add_codec_to_sdp(p, pref_codec, 8000, add_codec_to_sdp(p, pref_codec, 8000,
&m_audio_next, &m_audio_left, &m_audio_next, &m_audio_left,
&a_audio_next, &a_audio_left, &a_audio_next, &a_audio_left,
debug); debug);
else
add_codec_to_sdp(p, pref_codec, 90000,
&m_video_next, &m_video_left,
&a_video_next, &a_video_left,
debug);
alreadysent |= pref_codec; alreadysent |= pref_codec;
} }
/* Now send any other common codecs, and non-codec formats: */ /* Now send any other common audio and video codecs, and non-codec formats: */
for (x = 1; for (x = 1; x <= (needvideo ? AST_FORMAT_MAX_VIDEO : AST_FORMAT_MAX_AUDIO); x <<= 1) {
x <= ((ast_test_flag(&p->flags[1], SIP_PAGE2_VIDEOSUPPORT) && p->vrtp) ? AST_FORMAT_MAX_VIDEO : AST_FORMAT_MAX_AUDIO); if (!(capability & x)) /* Codec not requested */
x <<= 1) {
if (!(capability & x))
continue; continue;
if (alreadysent & x) ast_log(LOG_DEBUG, "--- Checking codec ... %d\n", x);
if (alreadysent & x) /* Already added to SDP */
continue; continue;
if (x <= AST_FORMAT_MAX_AUDIO) if (x <= AST_FORMAT_MAX_AUDIO)
@ -5085,6 +5275,7 @@ static int add_sdp(struct sip_request *resp, struct sip_pvt *p)
debug); debug);
} }
/* Now add DTMF RFC2833 telephony-event as a codec */
for (x = 1; x <= AST_RTP_MAX; x <<= 1) { for (x = 1; x <= AST_RTP_MAX; x <<= 1) {
if (!(p->noncodeccapability & x)) if (!(p->noncodeccapability & x))
continue; continue;
@ -5095,6 +5286,9 @@ static int add_sdp(struct sip_request *resp, struct sip_pvt *p)
debug); debug);
} }
if (option_debug > 2)
ast_log(LOG_DEBUG, "-- Done with adding codecs to SDP\n");
if(!ast_internal_timing_enabled(p->owner)) if(!ast_internal_timing_enabled(p->owner))
ast_build_string(&a_audio_next, &a_audio_left, "a=silenceSupp:off - - - -\r\n"); ast_build_string(&a_audio_next, &a_audio_left, "a=silenceSupp:off - - - -\r\n");
@ -5102,39 +5296,39 @@ static int add_sdp(struct sip_request *resp, struct sip_pvt *p)
ast_log(LOG_WARNING, "SIP SDP may be truncated due to undersized buffer!!\n"); ast_log(LOG_WARNING, "SIP SDP may be truncated due to undersized buffer!!\n");
ast_build_string(&m_audio_next, &m_audio_left, "\r\n"); ast_build_string(&m_audio_next, &m_audio_left, "\r\n");
if (needvideo)
ast_build_string(&m_video_next, &m_video_left, "\r\n"); ast_build_string(&m_video_next, &m_video_left, "\r\n");
len = strlen(v) + strlen(s) + strlen(o) + strlen(c) + strlen(t) + strlen(m_audio) + strlen(a_audio) + strlen(hold); len = strlen(version) + strlen(subject) + strlen(owner) + strlen(connection) + strlen(stime) + strlen(m_audio) + strlen(a_audio) + strlen(hold);
if ((p->vrtp) && if (needvideo) /* only if video response is appropriate */
(!ast_test_flag(&p->flags[0], SIP_NOVIDEO)) && len += strlen(m_video) + strlen(a_video) + strlen(bandwidth) + strlen(hold);
(capability & VIDEO_CODEC_MASK)) /* only if video response is appropriate */
len += strlen(m_video) + strlen(a_video) + strlen(b) + strlen(hold);
add_header(resp, "Content-Type", "application/sdp"); add_header(resp, "Content-Type", "application/sdp");
add_header_contentLength(resp, len); add_header_contentLength(resp, len);
add_line(resp, v); add_line(resp, version);
add_line(resp, o); add_line(resp, owner);
add_line(resp, s); add_line(resp, subject);
add_line(resp, c); add_line(resp, connection);
if ((p->vrtp) && if (needvideo) /* only if video response is appropriate */
(!ast_test_flag(&p->flags[0], SIP_NOVIDEO)) && add_line(resp, bandwidth);
(capability & VIDEO_CODEC_MASK)) /* only if video response is appropriate */ add_line(resp, stime);
add_line(resp, b);
add_line(resp, t);
add_line(resp, m_audio); add_line(resp, m_audio);
add_line(resp, a_audio); add_line(resp, a_audio);
add_line(resp, hold); add_line(resp, hold);
if ((p->vrtp) && if (needvideo) { /* only if video response is appropriate */
(!ast_test_flag(&p->flags[0], SIP_NOVIDEO)) &&
(capability & VIDEO_CODEC_MASK)) { /* only if video response is appropriate */
add_line(resp, m_video); add_line(resp, m_video);
add_line(resp, a_video); add_line(resp, a_video);
add_line(resp, hold); add_line(resp, hold); /* Repeat hold for the video stream */
} }
/* Update lastrtprx when we send our SDP */ /* Update lastrtprx when we send our SDP */
p->lastrtprx = p->lastrtptx = time(NULL); /* XXX why both ? */ p->lastrtprx = p->lastrtptx = time(NULL); /* XXX why both ? */
if (option_debug > 2) {
char buf[BUFSIZ];
ast_log(LOG_DEBUG, "Done building SDP. Settling with this capability: %s\n", ast_getformatname_multiple(buf, BUFSIZ, capability));
}
return 0; return 0;
} }
@ -7744,20 +7938,24 @@ static enum check_auth_result check_user_full(struct sip_pvt *p, struct sip_requ
p->pickupgroup = user->pickupgroup; p->pickupgroup = user->pickupgroup;
if (user->callingpres) /* User callingpres setting will override RPID header */ if (user->callingpres) /* User callingpres setting will override RPID header */
p->callingpres = user->callingpres; p->callingpres = user->callingpres;
p->capability = user->capability;
p->jointcapability = user->capability; /* Set default codec settings for this call */
p->maxcallbitrate = user->maxcallbitrate; p->capability = user->capability; /* User codec choice */
if (!ast_test_flag(&p->flags[1], SIP_PAGE2_VIDEOSUPPORT) && p->vrtp) { p->jointcapability = user->capability; /* Our codecs */
ast_rtp_destroy(p->vrtp); if (p->peercapability) /* AND with peer's codecs */
p->vrtp = NULL;
}
if (p->peercapability)
p->jointcapability &= p->peercapability; p->jointcapability &= p->peercapability;
if ((ast_test_flag(&p->flags[0], SIP_DTMF) == SIP_DTMF_RFC2833) || if ((ast_test_flag(&p->flags[0], SIP_DTMF) == SIP_DTMF_RFC2833) ||
(ast_test_flag(&p->flags[0], SIP_DTMF) == SIP_DTMF_AUTO)) (ast_test_flag(&p->flags[0], SIP_DTMF) == SIP_DTMF_AUTO))
p->noncodeccapability |= AST_RTP_DTMF; p->noncodeccapability |= AST_RTP_DTMF;
else else
p->noncodeccapability &= ~AST_RTP_DTMF; p->noncodeccapability &= ~AST_RTP_DTMF;
p->maxcallbitrate = user->maxcallbitrate;
/* If we do not support video, remove video from call structure */
if (!ast_test_flag(&p->flags[1], SIP_PAGE2_VIDEOSUPPORT) && p->vrtp) {
ast_rtp_destroy(p->vrtp);
p->vrtp = NULL;
}
} }
if (user && debug) if (user && debug)
ast_verbose("Found user '%s'\n", user->name); ast_verbose("Found user '%s'\n", user->name);
@ -10197,13 +10395,14 @@ static void check_pendings(struct sip_pvt *p)
} }
} }
/*! \brief Handle SIP response in dialogue */ /*! \brief Handle SIP response to INVITE dialogue */
static void handle_response_invite(struct sip_pvt *p, int resp, char *rest, struct sip_request *req, int seqno) static void handle_response_invite(struct sip_pvt *p, int resp, char *rest, struct sip_request *req, int seqno)
{ {
int outgoing = ast_test_flag(&p->flags[0], SIP_OUTGOING); int outgoing = ast_test_flag(&p->flags[0], SIP_OUTGOING);
int res = 0;
int reinvite = (p->owner && p->owner->_state == AST_STATE_UP);
if (option_debug > 3) { if (option_debug > 3) {
int reinvite = (p->owner && p->owner->_state == AST_STATE_UP);
if (reinvite) if (reinvite)
ast_log(LOG_DEBUG, "SIP response %d to RE-invite on %s call %s\n", resp, outgoing ? "outgoing" : "incoming", p->callid); ast_log(LOG_DEBUG, "SIP response %d to RE-invite on %s call %s\n", resp, outgoing ? "outgoing" : "incoming", p->callid);
else else
@ -10241,7 +10440,7 @@ static void handle_response_invite(struct sip_pvt *p, int resp, char *rest, stru
} }
} }
if (find_sdp(req)) { if (find_sdp(req)) {
process_sdp(p, req); res = process_sdp(p, req);
if (!ast_test_flag(req, SIP_PKT_IGNORE) && p->owner) { if (!ast_test_flag(req, SIP_PKT_IGNORE) && p->owner) {
/* Queue a progress frame only if we have SDP in 180 */ /* Queue a progress frame only if we have SDP in 180 */
ast_queue_control(p->owner, AST_CONTROL_PROGRESS); ast_queue_control(p->owner, AST_CONTROL_PROGRESS);
@ -10257,7 +10456,7 @@ static void handle_response_invite(struct sip_pvt *p, int resp, char *rest, stru
sip_cancel_destroy(p); sip_cancel_destroy(p);
/* Ignore 183 Session progress without SDP */ /* Ignore 183 Session progress without SDP */
if (find_sdp(req)) { if (find_sdp(req)) {
process_sdp(p, req); res = process_sdp(p, req);
if (!ast_test_flag(req, SIP_PKT_IGNORE) && p->owner) { if (!ast_test_flag(req, SIP_PKT_IGNORE) && p->owner) {
/* Queue a progress frame */ /* Queue a progress frame */
ast_queue_control(p->owner, AST_CONTROL_PROGRESS); ast_queue_control(p->owner, AST_CONTROL_PROGRESS);
@ -10272,8 +10471,13 @@ static void handle_response_invite(struct sip_pvt *p, int resp, char *rest, stru
if (!ast_test_flag(req, SIP_PKT_IGNORE)) if (!ast_test_flag(req, SIP_PKT_IGNORE))
sip_cancel_destroy(p); sip_cancel_destroy(p);
p->authtries = 0; p->authtries = 0;
if (find_sdp(req)) if (find_sdp(req)) {
process_sdp(p, req); if ((res = process_sdp(p, req)) && !ast_test_flag(req, SIP_PKT_IGNORE))
if (!reinvite)
/* This 200 OK's SDP is not acceptable, so we need to ack, then hangup */
/* For re-invites, we try to recover */
ast_set_flag(&p->flags[0], SIP_PENDINGBYE);
}
/* Parse contact header for continued conversation */ /* Parse contact header for continued conversation */
/* When we get 200 OK, we know which device (and IP) to contact for this call */ /* When we get 200 OK, we know which device (and IP) to contact for this call */
@ -10298,7 +10502,7 @@ static void handle_response_invite(struct sip_pvt *p, int resp, char *rest, stru
} }
if (!ast_test_flag(req, SIP_PKT_IGNORE) && p->owner) { if (!ast_test_flag(req, SIP_PKT_IGNORE) && p->owner) {
if (p->owner->_state != AST_STATE_UP) { if (!reinvite) {
ast_queue_control(p->owner, AST_CONTROL_ANSWER); ast_queue_control(p->owner, AST_CONTROL_ANSWER);
} else { /* RE-invite */ } else { /* RE-invite */
ast_queue_frame(p->owner, &ast_null_frame); ast_queue_frame(p->owner, &ast_null_frame);
@ -13460,6 +13664,9 @@ static struct ast_channel *sip_request_call(const char *type, int format, void *
*cause = AST_CAUSE_BEARERCAPABILITY_NOTAVAIL; /* Can't find codec to connect to host */ *cause = AST_CAUSE_BEARERCAPABILITY_NOTAVAIL; /* Can't find codec to connect to host */
return NULL; return NULL;
} }
if (option_debug)
ast_log(LOG_DEBUG, "Asked to create a SIP channel with formats: %s\n", ast_getformatname_multiple(tmp, sizeof(tmp), oldformat));
if (!(p = sip_alloc(NULL, NULL, 0, SIP_INVITE))) { if (!(p = sip_alloc(NULL, NULL, 0, SIP_INVITE))) {
ast_log(LOG_ERROR, "Unable to build sip pvt data for '%s' (Out of memory or socket error)\n", (char *)data); ast_log(LOG_ERROR, "Unable to build sip pvt data for '%s' (Out of memory or socket error)\n", (char *)data);
*cause = AST_CAUSE_SWITCH_CONGESTION; *cause = AST_CAUSE_SWITCH_CONGESTION;
@ -13509,7 +13716,7 @@ static struct ast_channel *sip_request_call(const char *type, int format, void *
#if 0 #if 0
printf("Setting up to call extension '%s' at '%s'\n", ext ? ext : "<none>", host); printf("Setting up to call extension '%s' at '%s'\n", ext ? ext : "<none>", host);
#endif #endif
p->prefcodec = format; p->prefcodec = oldformat; /* Format for this call */
ast_mutex_lock(&p->lock); ast_mutex_lock(&p->lock);
tmpc = sip_new(p, AST_STATE_DOWN, host); /* Place the call */ tmpc = sip_new(p, AST_STATE_DOWN, host); /* Place the call */
ast_mutex_unlock(&p->lock); ast_mutex_unlock(&p->lock);
@ -14744,10 +14951,13 @@ static int sip_set_rtp_peer(struct ast_channel *chan, struct ast_rtp *rtp, struc
changed = 1; changed = 1;
} }
if (changed && !ast_test_flag(&p->flags[0], SIP_GOTREFER)) { if (changed && !ast_test_flag(&p->flags[0], SIP_GOTREFER)) {
if (chan->_state != AST_STATE_UP) { if (chan->_state != AST_STATE_UP) { /* We are in early state */
char iabuf[INET_ADDRSTRLEN]; char iabuf[INET_ADDRSTRLEN];
ast_log(LOG_DEBUG, "Early media setting SIP '%s' - Sending early media to %s\n", p->callid, ast_inet_ntoa(iabuf, sizeof(iabuf), rtp ? p->redirip.sin_addr : p->ourip)); if (recordhistory)
} else if (!p->pendinginvite) { append_history(p, "ExtInv", "Initial invite sent with remote bridge proposal.");
if (option_debug)
ast_log(LOG_DEBUG, "Early remote bridge setting SIP '%s' - Sending media to %s\n", p->callid, ast_inet_ntoa(iabuf, sizeof(iabuf), rtp ? p->redirip.sin_addr : p->ourip));
} else if (!p->pendinginvite) { /* We are up, and have no outstanding invite */
if (option_debug > 2) { if (option_debug > 2) {
char iabuf[INET_ADDRSTRLEN]; char iabuf[INET_ADDRSTRLEN];
ast_log(LOG_DEBUG, "Sending reinvite on SIP '%s' - It's audio soon redirected to IP %s\n", p->callid, ast_inet_ntoa(iabuf, sizeof(iabuf), rtp ? p->redirip.sin_addr : p->ourip)); ast_log(LOG_DEBUG, "Sending reinvite on SIP '%s' - It's audio soon redirected to IP %s\n", p->callid, ast_inet_ntoa(iabuf, sizeof(iabuf), rtp ? p->redirip.sin_addr : p->ourip));
@ -14758,6 +14968,7 @@ static int sip_set_rtp_peer(struct ast_channel *chan, struct ast_rtp *rtp, struc
char iabuf[INET_ADDRSTRLEN]; char iabuf[INET_ADDRSTRLEN];
ast_log(LOG_DEBUG, "Deferring reinvite on SIP '%s' - It's audio will be redirected to IP %s\n", p->callid, ast_inet_ntoa(iabuf, sizeof(iabuf), rtp ? p->redirip.sin_addr : p->ourip)); ast_log(LOG_DEBUG, "Deferring reinvite on SIP '%s' - It's audio will be redirected to IP %s\n", p->callid, ast_inet_ntoa(iabuf, sizeof(iabuf), rtp ? p->redirip.sin_addr : p->ourip));
} }
/* We have a pending Invite. Send re-invite when we're done with the invite */
ast_set_flag(&p->flags[0], SIP_NEEDREINVITE); ast_set_flag(&p->flags[0], SIP_NEEDREINVITE);
} }
} }

10
frame.c
Unescape View File

@ -1008,9 +1008,12 @@ int ast_codec_choose(struct ast_codec_pref *pref, int formats, int find_best)
break; break;
} }
} }
if(ret) if(ret & AST_FORMAT_AUDIO_MASK)
return ret; return ret;
if (option_debug > 3)
ast_log(LOG_DEBUG, "Could not find preferred codec - %s\n", find_best ? "Going for the best codec" : "Returning zero codec");
return find_best ? ast_best_codec(formats) : 0; return find_best ? ast_best_codec(formats) : 0;
} }
@ -1034,7 +1037,10 @@ void ast_parse_allow_disallow(struct ast_codec_pref *pref, int *mask, const char
*mask &= ~format; *mask &= ~format;
} }
if (pref) { /* Set up a preference list for audio. Do not include video in preferences
since we can not transcode video and have to use whatever is offered
*/
if (pref && (format & AST_FORMAT_AUDIO_MASK)) {
if (strcasecmp(this, "all")) { if (strcasecmp(this, "all")) {
if (allowing) if (allowing)
ast_codec_pref_append(pref, format); ast_codec_pref_append(pref, format);

4
include/asterisk/rtp.h
Unescape View File

@ -62,6 +62,9 @@ struct ast_rtp_protocol {
AST_LIST_ENTRY(ast_rtp_protocol) list; AST_LIST_ENTRY(ast_rtp_protocol) list;
}; };
#define FLAG_3389_WARNING (1 << 0)
typedef int (*ast_rtp_callback)(struct ast_rtp *rtp, struct ast_frame *f, void *data); typedef int (*ast_rtp_callback)(struct ast_rtp *rtp, struct ast_frame *f, void *data);
@ -71,7 +74,6 @@ typedef int (*ast_rtp_callback)(struct ast_rtp *rtp, struct ast_frame *f, void *
* RTP session is defined on page 9 of RFC 3550: "An association among a set of participants communicating with RTP. A participant may be involved in multiple RTP sessions at the same time [...]" * RTP session is defined on page 9 of RFC 3550: "An association among a set of participants communicating with RTP. A participant may be involved in multiple RTP sessions at the same time [...]"
* *
*/ */
/*! \brief The value of each payload format mapping: */ /*! \brief The value of each payload format mapping: */
struct rtpPayloadType { struct rtpPayloadType {
int isAstFormat; /*!< whether the following code is an AST_FORMAT */ int isAstFormat; /*!< whether the following code is an AST_FORMAT */

3
include/asterisk/translate.h
Unescape View File

@ -23,7 +23,8 @@
#ifndef _ASTERISK_TRANSLATE_H #ifndef _ASTERISK_TRANSLATE_H
#define _ASTERISK_TRANSLATE_H #define _ASTERISK_TRANSLATE_H
#define MAX_FORMAT 32 //#define MAX_FORMAT 15 /* Do not include video here */
#define MAX_FORMAT 32 /* Do include video here */
#if defined(__cplusplus) || defined(c_plusplus) #if defined(__cplusplus) || defined(c_plusplus)
extern "C" { extern "C" {

7
rtp.c
Unescape View File

@ -1350,7 +1350,7 @@ int ast_rtp_make_compatible(struct ast_channel *dest, struct ast_channel *src, i
return 1; return 1;
} }
/*! \brief Make a note of a RTP paymoad type that was seen in a SDP "m=" line. /*! \brief Make a note of a RTP payload type that was seen in a SDP "m=" line.
* By default, use the well-known value for this type (although it may * By default, use the well-known value for this type (although it may
* still be set to a different value by a subsequent "a=rtpmap:" line) * still be set to a different value by a subsequent "a=rtpmap:" line)
*/ */
@ -1359,10 +1359,9 @@ void ast_rtp_set_m_type(struct ast_rtp* rtp, int pt)
if (pt < 0 || pt > MAX_RTP_PT) if (pt < 0 || pt > MAX_RTP_PT)
return; /* bogus payload type */ return; /* bogus payload type */
if (static_RTP_PT[pt].code != 0) { if (static_RTP_PT[pt].code != 0)
rtp->current_RTP_PT[pt] = static_RTP_PT[pt]; rtp->current_RTP_PT[pt] = static_RTP_PT[pt];
} }
}
/*! \brief Make a note of a RTP payload type (with MIME type) that was seen in /*! \brief Make a note of a RTP payload type (with MIME type) that was seen in
a SDP "a=rtpmap:" line. */ a SDP "a=rtpmap:" line. */
@ -2245,7 +2244,7 @@ int ast_rtp_write(struct ast_rtp *rtp, struct ast_frame *_f)
/* Make sure we have enough space for RTP header */ /* Make sure we have enough space for RTP header */
if ((_f->frametype != AST_FRAME_VOICE) && (_f->frametype != AST_FRAME_VIDEO)) { if ((_f->frametype != AST_FRAME_VOICE) && (_f->frametype != AST_FRAME_VIDEO)) {
ast_log(LOG_WARNING, "RTP can only send voice\n"); ast_log(LOG_WARNING, "RTP can only send voice and video\n");
return -1; return -1;
} }

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