diff options
Diffstat (limited to 'channels/chan_sip.c')
| -rw-r--r-- | channels/chan_sip.c | 791 |
1 files changed, 412 insertions, 379 deletions
diff --git a/channels/chan_sip.c b/channels/chan_sip.c index 095b5a763..44629dc3d 100644 --- a/channels/chan_sip.c +++ b/channels/chan_sip.c @@ -285,6 +285,7 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$") #include "asterisk/stasis_channels.h" #include "asterisk/features_config.h" #include "asterisk/http_websocket.h" +#include "asterisk/format_cache.h" /*** DOCUMENTATION <application name="SIPDtmfMode" language="en_US"> @@ -747,7 +748,6 @@ static char default_mohsuggest[MAX_MUSICCLASS]; /*!< Global setting for moh c static char default_parkinglot[AST_MAX_CONTEXT]; /*!< Parkinglot */ static char default_engine[256]; /*!< Default RTP engine */ static int default_maxcallbitrate; /*!< Maximum bitrate for call */ -static struct ast_codec_pref default_prefs; /*!< Default codec prefs */ static char default_zone[MAX_TONEZONE_COUNTRY]; /*!< Default tone zone for channels created from the SIP driver */ static unsigned int default_transports; /*!< Default Transports (enum ast_transport) that are acceptable */ static unsigned int default_primary_transport; /*!< Default primary Transport (enum ast_transport) for outbound connections to devices */ @@ -1295,7 +1295,6 @@ static int str2dtmfmode(const char *str) attribute_unused; static const char *insecure2str(int mode) attribute_const; static const char *allowoverlap2str(int mode) attribute_const; static void cleanup_stale_contexts(char *new, char *old); -static void print_codec_to_cli(int fd, struct ast_codec_pref *pref); static const char *domain_mode_to_text(const enum domain_mode mode); static char *sip_show_domains(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a); static char *_sip_show_peer(int type, int fd, struct mansession *s, const struct message *m, int argc, const char *argv[]); @@ -1764,7 +1763,7 @@ static void destroy_escs(void) { int i; for (i = 0; i < ARRAY_LEN(event_state_compositors); i++) { - ao2_ref(event_state_compositors[i].compositor, -1); + ao2_cleanup(event_state_compositors[i].compositor); } } @@ -3385,7 +3384,7 @@ static void unlink_peers_from_tables(peer_unlink_flag_t flag) ao2_t_callback(peers, OBJ_NODATA | OBJ_UNLINK | OBJ_MULTIPLE, match_and_cleanup_peer_sched, &flag, "initiating callback to remove marked peers"); ao2_t_callback(peers_by_ip, OBJ_NODATA | OBJ_UNLINK | OBJ_MULTIPLE, - match_and_cleanup_peer_sched, &flag, "initiating callback to remove marked peers"); + match_and_cleanup_peer_sched, &flag, "initiating callback to remove marked peers_by_ip"); } /* \brief Unlink all marked peers from ao2 containers */ @@ -5308,7 +5307,7 @@ static void sip_destroy_peer(struct sip_peer *peer) ast_string_field_free_memory(peer); - peer->caps = ast_format_cap_destroy(peer->caps); + ao2_cleanup(peer->caps); ast_rtp_dtls_cfg_free(&peer->dtls_cfg); @@ -5967,7 +5966,7 @@ static int dialog_initialize_rtp(struct sip_pvt *dialog) } if (ast_test_flag(&dialog->flags[1], SIP_PAGE2_VIDEOSUPPORT_ALWAYS) || - (ast_test_flag(&dialog->flags[1], SIP_PAGE2_VIDEOSUPPORT) && (ast_format_cap_has_type(dialog->caps, AST_FORMAT_TYPE_VIDEO)))) { + (ast_test_flag(&dialog->flags[1], SIP_PAGE2_VIDEOSUPPORT) && (ast_format_cap_has_type(dialog->caps, AST_MEDIA_TYPE_VIDEO)))) { if (!(dialog->vrtp = ast_rtp_instance_new(dialog->engine, sched, &bindaddr_tmp, NULL))) { return -1; } @@ -6053,8 +6052,11 @@ static int create_addr_from_peer(struct sip_pvt *dialog, struct sip_peer *peer) ast_copy_flags(&dialog->flags[0], &peer->flags[0], SIP_FLAGS_TO_COPY); ast_copy_flags(&dialog->flags[1], &peer->flags[1], SIP_PAGE2_FLAGS_TO_COPY); ast_copy_flags(&dialog->flags[2], &peer->flags[2], SIP_PAGE3_FLAGS_TO_COPY); - ast_format_cap_copy(dialog->caps, peer->caps); - dialog->prefs = peer->prefs; + /* Take the peer's caps */ + if (peer->caps) { + ast_format_cap_remove_by_type(dialog->caps, AST_MEDIA_TYPE_UNKNOWN); + ast_format_cap_append_from_cap(dialog->caps, peer->caps, AST_MEDIA_TYPE_UNKNOWN); + } dialog->amaflags = peer->amaflags; ast_string_field_set(dialog, engine, peer->engine); @@ -6078,8 +6080,8 @@ static int create_addr_from_peer(struct sip_pvt *dialog, struct sip_peer *peer) ast_rtp_instance_set_prop(dialog->rtp, AST_RTP_PROPERTY_DTMF, ast_test_flag(&dialog->flags[0], SIP_DTMF) == SIP_DTMF_RFC2833); ast_rtp_instance_set_prop(dialog->rtp, AST_RTP_PROPERTY_DTMF_COMPENSATE, ast_test_flag(&dialog->flags[1], SIP_PAGE2_RFC2833_COMPENSATE)); /* Set Frame packetization */ - ast_rtp_codecs_packetization_set(ast_rtp_instance_get_codecs(dialog->rtp), dialog->rtp, &dialog->prefs); dialog->autoframing = peer->autoframing; + ast_rtp_codecs_set_framing(ast_rtp_instance_get_codecs(dialog->rtp), ast_format_cap_get_framing(dialog->caps)); } /* XXX TODO: get fields directly from peer only as they are needed using dialog->relatedpeer */ @@ -6456,7 +6458,7 @@ static int sip_call(struct ast_channel *ast, const char *dest, int timeout) p->jointnoncodeccapability = p->noncodeccapability; /* If there are no audio formats left to offer, punt */ - if (!(ast_format_cap_has_type(p->jointcaps, AST_FORMAT_TYPE_AUDIO))) { + if (!(ast_format_cap_has_type(p->jointcaps, AST_MEDIA_TYPE_AUDIO))) { ast_log(LOG_WARNING, "No audio format found to offer. Cancelling call to %s\n", p->username); res = -1; } else { @@ -6721,11 +6723,11 @@ void __sip_destroy(struct sip_pvt *p, int lockowner, int lockdialoglist) p->named_callgroups = ast_unref_namedgroups(p->named_callgroups); p->named_pickupgroups = ast_unref_namedgroups(p->named_pickupgroups); - p->caps = ast_format_cap_destroy(p->caps); - p->jointcaps = ast_format_cap_destroy(p->jointcaps); - p->peercaps = ast_format_cap_destroy(p->peercaps); - p->redircaps = ast_format_cap_destroy(p->redircaps); - p->prefcaps = ast_format_cap_destroy(p->prefcaps); + ao2_cleanup(p->caps); + ao2_cleanup(p->jointcaps); + ao2_cleanup(p->peercaps); + ao2_cleanup(p->redircaps); + ao2_cleanup(p->prefcaps); ast_rtp_dtls_cfg_free(&p->dtls_cfg); @@ -7308,7 +7310,6 @@ static int sip_hangup(struct ast_channel *ast) /*! \brief Try setting the codecs suggested by the SIP_CODEC channel variable */ static void try_suggested_sip_codec(struct sip_pvt *p) { - struct ast_format fmt; const char *codec_list; char *codec_list_copy; struct ast_format_cap *original_jointcaps; @@ -7328,31 +7329,44 @@ static void try_suggested_sip_codec(struct sip_pvt *p) } codec_list_copy = ast_strdupa(codec_list); - original_jointcaps = ast_format_cap_dup(p->jointcaps); + + original_jointcaps = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT); + if (!original_jointcaps) { + return; + } + ast_format_cap_append_from_cap(original_jointcaps, p->jointcaps, AST_MEDIA_TYPE_UNKNOWN); for (codec = strtok_r(codec_list_copy, ",", &strtok_ptr); codec; codec = strtok_r(NULL, ",", &strtok_ptr)) { + struct ast_format *fmt; + codec = ast_strip(codec); - if (!ast_getformatbyname(codec, &fmt)) { + fmt = ast_format_cache_get(codec); + if (!fmt) { ast_log(AST_LOG_NOTICE, "Ignoring ${SIP_CODEC*} variable because of unrecognized/not configured codec %s (check allow/disallow in sip.conf)\n", codec); continue; } - if (ast_format_cap_iscompatible(original_jointcaps, &fmt)) { + if (ast_format_cap_iscompatible_format(original_jointcaps, fmt) != AST_FORMAT_CMP_NOT_EQUAL) { if (first_codec) { ast_verb(4, "Set codec to '%s' for this call because of ${SIP_CODEC*} variable\n", codec); - ast_format_cap_set(p->jointcaps, &fmt); - ast_format_cap_set(p->caps, &fmt); + ast_format_cap_remove_by_type(p->jointcaps, AST_MEDIA_TYPE_UNKNOWN); + ast_format_cap_append(p->jointcaps, fmt, 0); + ast_format_cap_remove_by_type(p->caps, AST_MEDIA_TYPE_UNKNOWN); + ast_format_cap_append(p->caps, fmt, 0); first_codec = 0; } else { ast_verb(4, "Add codec to '%s' for this call because of ${SIP_CODEC*} variable\n", codec); - ast_format_cap_add(p->jointcaps, &fmt); - ast_format_cap_add(p->caps, &fmt); + /* Add the format to the capabilities structure */ + ast_format_cap_append(p->jointcaps, fmt, 0); + ast_format_cap_append(p->caps, fmt, 0); } } else { ast_log(AST_LOG_NOTICE, "Ignoring ${SIP_CODEC*} variable because it is not shared by both ends: %s\n", codec); } + + ao2_ref(fmt, -1); } - ast_format_cap_destroy(original_jointcaps); + ao2_ref(original_jointcaps, -1); return; } @@ -7401,13 +7415,13 @@ static int sip_write(struct ast_channel *ast, struct ast_frame *frame) switch (frame->frametype) { case AST_FRAME_VOICE: - if (!(ast_format_cap_iscompatible(ast_channel_nativeformats(ast), &frame->subclass.format))) { - char s1[512]; + if (ast_format_cap_iscompatible_format(ast_channel_nativeformats(ast), frame->subclass.format) == AST_FORMAT_CMP_NOT_EQUAL) { + struct ast_str *codec_buf = ast_str_alloca(64); ast_log(LOG_WARNING, "Asked to transmit frame type %s, while native formats is %s read/write = %s/%s\n", - ast_getformatname(&frame->subclass.format), - ast_getformatname_multiple(s1, sizeof(s1), ast_channel_nativeformats(ast)), - ast_getformatname(ast_channel_readformat(ast)), - ast_getformatname(ast_channel_writeformat(ast))); + ast_format_get_name(frame->subclass.format), + ast_format_cap_get_names(ast_channel_nativeformats(ast), &codec_buf), + ast_format_get_name(ast_channel_readformat(ast)), + ast_format_get_name(ast_channel_writeformat(ast))); return 0; } if (p) { @@ -7929,10 +7943,7 @@ static int sip_indicate(struct ast_channel *ast, int condition, const void *data if (p->vrtp && !p->novideo) { /* FIXME: Only use this for VP8. Additional work would have to be done to * fully support other video codecs */ - struct ast_format_cap *fcap = ast_channel_nativeformats(ast); - struct ast_format vp8; - ast_format_set(&vp8, AST_FORMAT_VP8, 0); - if (ast_format_cap_iscompatible(fcap, &vp8)) { + if (ast_format_cap_iscompatible_format(ast_channel_nativeformats(ast), ast_format_vp8) != AST_FORMAT_CMP_NOT_EQUAL) { /* FIXME Fake RTP write, this will be sent as an RTCP packet. Ideally the * RTP engine would provide a way to externally write/schedule RTCP * packets */ @@ -8036,15 +8047,21 @@ static int sip_indicate(struct ast_channel *ast, int condition, const void *data */ static struct ast_channel *sip_new(struct sip_pvt *i, int state, const char *title, const struct ast_assigned_ids *assignedids, const struct ast_channel *requestor, struct ast_callid *callid) { + struct ast_format_cap *caps; struct ast_channel *tmp; struct ast_variable *v = NULL; - struct ast_format fmt; + struct ast_format *fmt; struct ast_format_cap *what = NULL; /* SHALLOW COPY DO NOT DESTROY! */ + struct ast_str *codec_buf = ast_str_alloca(64); int needvideo = 0; int needtext = 0; - char buf[SIPBUFSIZE]; char *exten; + caps = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT); + if (!caps) { + return NULL; + } + { const char *my_name; /* pick a good name */ @@ -8060,6 +8077,7 @@ static struct ast_channel *sip_new(struct sip_pvt *i, int state, const char *tit } if (!tmp) { ast_log(LOG_WARNING, "Unable to allocate AST channel structure for SIP channel\n"); + ao2_ref(caps, -1); sip_pvt_lock(i); return NULL; } @@ -8068,6 +8086,7 @@ static struct ast_channel *sip_new(struct sip_pvt *i, int state, const char *tit if (ast_endpoint_add_channel(i->relatedpeer->endpoint, tmp)) { ast_channel_unlock(tmp); ast_channel_unref(tmp); + ao2_ref(caps, -1); sip_pvt_lock(i); return NULL; } @@ -8088,27 +8107,40 @@ static struct ast_channel *sip_new(struct sip_pvt *i, int state, const char *tit /* Select our native format based on codec preference until we receive something from another device to the contrary. */ - if (!(ast_format_cap_is_empty(i->jointcaps))) { /* The joint capabilities of us and peer */ + if (ast_format_cap_count(i->jointcaps)) { /* The joint capabilities of us and peer */ what = i->jointcaps; - } else if (!(ast_format_cap_is_empty(i->caps))) { /* Our configured capability for this peer */ + } else if (ast_format_cap_count(i->caps)) { /* Our configured capability for this peer */ what = i->caps; } else { what = sip_cfg.caps; } /* Set the native formats */ - ast_format_cap_copy(ast_channel_nativeformats(tmp), what); - /* choose and use only the best audio format for our native formats */ - ast_codec_choose(&i->prefs, ast_channel_nativeformats(tmp), 1, &fmt); /* get the best audio format */ - ast_format_cap_remove_bytype(ast_channel_nativeformats(tmp), AST_FORMAT_TYPE_AUDIO); /* remove only the other audio formats */ - ast_format_cap_add(ast_channel_nativeformats(tmp), &fmt); /* add our best choice back */ + ast_format_cap_append_from_cap(caps, what, AST_MEDIA_TYPE_UNKNOWN); + /* Use only the preferred audio format, which is stored at the '0' index */ + fmt = ast_format_cap_get_best_by_type(what, AST_MEDIA_TYPE_AUDIO); /* get the best audio format */ + if (fmt) { + ast_format_cap_remove_by_type(caps, AST_MEDIA_TYPE_AUDIO); /* remove only the other audio formats */ + ast_format_cap_append(caps, fmt, 0); /* add our best choice back */ + } else { + /* If we don't have an audio format, try to get something */ + fmt = ast_format_cap_get_format(caps, 0); + if (!fmt) { + ast_log(LOG_WARNING, "No compatible formats could be found for %s\n", ast_channel_name(tmp)); + ao2_ref(caps, -1); + tmp = ast_channel_unref(tmp); + return NULL; + } + } + ast_channel_nativeformats_set(tmp, caps); + ao2_ref(caps, -1); - ast_debug(3, "*** Our native formats are %s \n", ast_getformatname_multiple(buf, SIPBUFSIZE, ast_channel_nativeformats(tmp))); - ast_debug(3, "*** Joint capabilities are %s \n", ast_getformatname_multiple(buf, SIPBUFSIZE, i->jointcaps)); - ast_debug(3, "*** Our capabilities are %s \n", ast_getformatname_multiple(buf, SIPBUFSIZE, i->caps)); - ast_debug(3, "*** AST_CODEC_CHOOSE formats are %s \n", ast_getformatname(&fmt)); - if (!ast_format_cap_is_empty(i->prefcaps)) { - ast_debug(3, "*** Our preferred formats from the incoming channel are %s \n", ast_getformatname_multiple(buf, SIPBUFSIZE, i->prefcaps)); + ast_debug(3, "*** Our native formats are %s \n", ast_format_cap_get_names(ast_channel_nativeformats(tmp), &codec_buf)); + ast_debug(3, "*** Joint capabilities are %s \n", ast_format_cap_get_names(i->jointcaps, &codec_buf)); + ast_debug(3, "*** Our capabilities are %s \n", ast_format_cap_get_names(i->caps, &codec_buf)); + ast_debug(3, "*** AST_CODEC_CHOOSE formats are %s \n", ast_format_get_name(fmt)); + if (ast_format_cap_count(i->prefcaps)) { + ast_debug(3, "*** Our preferred formats from the incoming channel are %s \n", ast_format_cap_get_names(i->prefcaps, &codec_buf)); } /* If we have a prefcodec setting, we have an inbound channel that set a @@ -8118,17 +8150,17 @@ static struct ast_channel *sip_new(struct sip_pvt *i, int state, const char *tit if (i->vrtp) { if (ast_test_flag(&i->flags[1], SIP_PAGE2_VIDEOSUPPORT)) needvideo = 1; - else if (!ast_format_cap_is_empty(i->prefcaps)) - needvideo = ast_format_cap_has_type(i->prefcaps, AST_FORMAT_TYPE_VIDEO); /* Outbound call */ + else if (ast_format_cap_count(i->prefcaps)) + needvideo = ast_format_cap_has_type(i->prefcaps, AST_MEDIA_TYPE_VIDEO); /* Outbound call */ else - needvideo = ast_format_cap_has_type(i->jointcaps, AST_FORMAT_TYPE_VIDEO); /* Inbound call */ + needvideo = ast_format_cap_has_type(i->jointcaps, AST_MEDIA_TYPE_VIDEO); /* Inbound call */ } if (i->trtp) { - if (!ast_format_cap_is_empty(i->prefcaps)) - needtext = ast_format_cap_has_type(i->prefcaps, AST_FORMAT_TYPE_TEXT); /* Outbound call */ + if (ast_format_cap_count(i->prefcaps)) + needtext = ast_format_cap_has_type(i->prefcaps, AST_MEDIA_TYPE_TEXT); /* Outbound call */ else - needtext = ast_format_cap_has_type(i->jointcaps, AST_FORMAT_TYPE_TEXT); /* Inbound call */ + needtext = ast_format_cap_has_type(i->jointcaps, AST_MEDIA_TYPE_TEXT); /* Inbound call */ } if (needvideo) { @@ -8156,8 +8188,8 @@ static struct ast_channel *sip_new(struct sip_pvt *i, int state, const char *tit if (i->rtp) { ast_channel_set_fd(tmp, 0, ast_rtp_instance_fd(i->rtp, 0)); ast_channel_set_fd(tmp, 1, ast_rtp_instance_fd(i->rtp, 1)); - ast_rtp_instance_set_write_format(i->rtp, &fmt); - ast_rtp_instance_set_read_format(i->rtp, &fmt); + ast_rtp_instance_set_write_format(i->rtp, fmt); + ast_rtp_instance_set_read_format(i->rtp, fmt); } if (needvideo && i->vrtp) { ast_channel_set_fd(tmp, 2, ast_rtp_instance_fd(i->vrtp, 0)); @@ -8175,11 +8207,13 @@ static struct ast_channel *sip_new(struct sip_pvt *i, int state, const char *tit } ast_channel_adsicpe_set(tmp, AST_ADSI_UNAVAILABLE); - ast_format_copy(ast_channel_writeformat(tmp), &fmt); - ast_format_copy(ast_channel_rawwriteformat(tmp), &fmt); + ast_channel_set_writeformat(tmp, fmt); + ast_channel_set_rawwriteformat(tmp, fmt); + + ast_channel_set_readformat(tmp, fmt); + ast_channel_set_rawreadformat(tmp, fmt); - ast_format_copy(ast_channel_readformat(tmp), &fmt); - ast_format_copy(ast_channel_rawreadformat(tmp), &fmt); + ao2_ref(fmt, -1); ast_channel_tech_pvt_set(tmp, dialog_ref(i, "sip_new: set chan->tech_pvt to i")); @@ -8534,17 +8568,26 @@ static struct ast_frame *sip_rtp_read(struct ast_channel *ast, struct sip_pvt *p return f; } - if (f && !ast_format_cap_iscompatible(ast_channel_nativeformats(p->owner), &f->subclass.format)) { - if (!ast_format_cap_iscompatible(p->jointcaps, &f->subclass.format)) { + if (f && ast_format_cap_iscompatible_format(ast_channel_nativeformats(p->owner), f->subclass.format) == AST_FORMAT_CMP_NOT_EQUAL) { + struct ast_format_cap *caps; + + if (ast_format_cap_iscompatible_format(p->jointcaps, f->subclass.format) == AST_FORMAT_CMP_NOT_EQUAL) { ast_debug(1, "Bogus frame of format '%s' received from '%s'!\n", - ast_getformatname(&f->subclass.format), ast_channel_name(p->owner)); + ast_format_get_name(f->subclass.format), ast_channel_name(p->owner)); ast_frfree(f); return &ast_null_frame; } ast_debug(1, "Oooh, format changed to %s\n", - ast_getformatname(&f->subclass.format)); - ast_format_cap_remove_bytype(ast_channel_nativeformats(p->owner), AST_FORMAT_TYPE_AUDIO); - ast_format_cap_add(ast_channel_nativeformats(p->owner), &f->subclass.format); + ast_format_get_name(f->subclass.format)); + + caps = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT); + if (caps) { + ast_format_cap_append_from_cap(caps, ast_channel_nativeformats(p->owner), AST_MEDIA_TYPE_UNKNOWN); + ast_format_cap_remove_by_type(caps, AST_MEDIA_TYPE_AUDIO); + ast_format_cap_append(caps, f->subclass.format, 0); + ast_channel_nativeformats_set(p->owner, caps); + ao2_ref(caps, -1); + } ast_set_read_format(p->owner, ast_channel_readformat(p->owner)); ast_set_write_format(p->owner, ast_channel_writeformat(p->owner)); } @@ -8796,18 +8839,18 @@ struct sip_pvt *sip_alloc(ast_string_field callid, struct ast_sockaddr *addr, sip_pvt_callid_set(p, logger_callid); } - p->caps = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_NOLOCK); - p->jointcaps = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_NOLOCK); - p->peercaps = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_NOLOCK); - p->redircaps = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_NOLOCK); - p->prefcaps = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_NOLOCK); + p->caps = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT); + p->jointcaps = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT); + p->peercaps = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT); + p->redircaps = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT); + p->prefcaps = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT); - if (!p->caps|| !p->jointcaps || !p->peercaps || !p->redircaps) { - p->caps = ast_format_cap_destroy(p->caps); - p->jointcaps = ast_format_cap_destroy(p->jointcaps); - p->peercaps = ast_format_cap_destroy(p->peercaps); - p->redircaps = ast_format_cap_destroy(p->redircaps); - p->prefcaps = ast_format_cap_destroy(p->prefcaps); + if (!p->caps|| !p->jointcaps || !p->peercaps || !p->redircaps || !p->prefcaps) { + ao2_cleanup(p->caps); + ao2_cleanup(p->jointcaps); + ao2_cleanup(p->peercaps); + ao2_cleanup(p->redircaps); + ao2_cleanup(p->prefcaps); ao2_t_ref(p, -1, "Yuck, couldn't allocate format capabilities. Get rid o' p"); return NULL; } @@ -8857,7 +8900,6 @@ struct sip_pvt *sip_alloc(ast_string_field callid, struct ast_sockaddr *addr, p->sessionversion_remote = -1; p->session_modify = TRUE; p->stimer = NULL; - p->prefs = default_prefs; /* Set default codecs for this call */ ast_copy_string(p->zone, default_zone, sizeof(p->zone)); p->maxforwards = sip_cfg.default_max_forwards; @@ -8909,7 +8951,7 @@ struct sip_pvt *sip_alloc(ast_string_field callid, struct ast_sockaddr *addr, /* Assign default music on hold class */ ast_string_field_set(p, mohinterpret, default_mohinterpret); ast_string_field_set(p, mohsuggest, default_mohsuggest); - ast_format_cap_copy(p->caps, sip_cfg.caps); + ast_format_cap_append_from_cap(p->caps, sip_cfg.caps, AST_MEDIA_TYPE_UNKNOWN); p->allowtransfer = sip_cfg.allowtransfer; if ((ast_test_flag(&p->flags[0], SIP_DTMF) == SIP_DTMF_RFC2833) || (ast_test_flag(&p->flags[0], SIP_DTMF) == SIP_DTMF_AUTO)) { @@ -10034,15 +10076,17 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action int udptlportno = -1; /*!< UDPTL image destination port number */ /* Peer capability is the capability in the SDP, non codec is RFC2833 DTMF (101) */ - struct ast_format_cap *peercapability = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_NOLOCK); - struct ast_format_cap *vpeercapability = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_NOLOCK); - struct ast_format_cap *tpeercapability = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_NOLOCK); + struct ast_format_cap *peercapability = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT); + struct ast_format_cap *vpeercapability = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT); + struct ast_format_cap *tpeercapability = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT); int peernoncodeccapability = 0, vpeernoncodeccapability = 0, tpeernoncodeccapability = 0; - struct ast_rtp_codecs newaudiortp = { 0, }, newvideortp = { 0, }, newtextrtp = { 0, }; - struct ast_format_cap *newjointcapability = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_NOLOCK); /* Negotiated capability */ - struct ast_format_cap *newpeercapability = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_NOLOCK); + struct ast_rtp_codecs newaudiortp = AST_RTP_CODECS_NULL_INIT; + struct ast_rtp_codecs newvideortp = AST_RTP_CODECS_NULL_INIT; + struct ast_rtp_codecs newtextrtp = AST_RTP_CODECS_NULL_INIT; + struct ast_format_cap *newjointcapability = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT); /* Negotiated capability */ + struct ast_format_cap *newpeercapability = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT); int newnoncodeccapability; const char *codecs; @@ -10062,8 +10106,8 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action int debug = sip_debug_test_pvt(p); /* START UNKNOWN */ - char buf[SIPBUFSIZE]; - struct ast_format tmp_fmt; + struct ast_str *codec_buf = ast_str_alloca(64); + struct ast_format *tmp_fmt; /* END UNKNOWN */ /* Initial check */ @@ -10625,10 +10669,12 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action /* Ensure crypto lines are provided where necessary */ if (audio && secure_audio && !processed_crypto) { ast_log(LOG_WARNING, "Rejecting secure audio stream without encryption details: %s\n", m); - return -1; + res = -1; + goto process_sdp_cleanup; } else if (video && secure_video && !processed_crypto) { ast_log(LOG_WARNING, "Rejecting secure video stream without encryption details: %s\n", m); - return -1; + res = -1; + goto process_sdp_cleanup; } } @@ -10687,12 +10733,12 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action ast_rtp_codecs_payload_formats(&newvideortp, vpeercapability, &vpeernoncodeccapability); ast_rtp_codecs_payload_formats(&newtextrtp, tpeercapability, &tpeernoncodeccapability); - ast_format_cap_append(newpeercapability, peercapability); - ast_format_cap_append(newpeercapability, vpeercapability); - ast_format_cap_append(newpeercapability, tpeercapability); + ast_format_cap_append_from_cap(newpeercapability, peercapability, AST_MEDIA_TYPE_AUDIO); + ast_format_cap_append_from_cap(newpeercapability, vpeercapability, AST_MEDIA_TYPE_VIDEO); + ast_format_cap_append_from_cap(newpeercapability, tpeercapability, AST_MEDIA_TYPE_TEXT); - ast_format_cap_joint_copy(p->caps, newpeercapability, newjointcapability); - if (ast_format_cap_is_empty(newjointcapability) && udptlportno == -1) { + ast_format_cap_get_compatible(p->caps, newpeercapability, newjointcapability); + if (!ast_format_cap_count(newjointcapability) && udptlportno == -1) { ast_log(LOG_NOTICE, "No compatible codecs, not accepting this offer!\n"); /* Do NOT Change current setting */ res = -1; @@ -10703,14 +10749,18 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action if (debug) { /* shame on whoever coded this.... */ - char s1[SIPBUFSIZE], s2[SIPBUFSIZE], s3[SIPBUFSIZE], s4[SIPBUFSIZE], s5[SIPBUFSIZE]; + struct ast_str *cap_buf = ast_str_alloca(64); + struct ast_str *peer_buf = ast_str_alloca(64); + struct ast_str *vpeer_buf = ast_str_alloca(64); + struct ast_str *tpeer_buf = ast_str_alloca(64); + struct ast_str *joint_buf = ast_str_alloca(64); ast_verbose("Capabilities: us - %s, peer - audio=%s/video=%s/text=%s, combined - %s\n", - ast_getformatname_multiple(s1, SIPBUFSIZE, p->caps), - ast_getformatname_multiple(s2, SIPBUFSIZE, peercapability), - ast_getformatname_multiple(s3, SIPBUFSIZE, vpeercapability), - ast_getformatname_multiple(s4, SIPBUFSIZE, tpeercapability), - ast_getformatname_multiple(s5, SIPBUFSIZE, newjointcapability)); + ast_format_cap_get_names(p->caps, &cap_buf), + ast_format_cap_get_names(peercapability, &peer_buf), + ast_format_cap_get_names(vpeercapability, &vpeer_buf), + ast_format_cap_get_names(tpeercapability, &tpeer_buf), + ast_format_cap_get_names(newjointcapability, &joint_buf)); } if (debug) { struct ast_str *s1 = ast_str_alloca(SIPBUFSIZE); @@ -10726,14 +10776,21 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action if (portno != -1 || vportno != -1 || tportno != -1) { /* We are now ready to change the sip session and RTP structures with the offered codecs, since they are acceptable */ - ast_format_cap_copy(p->jointcaps, newjointcapability); /* Our joint codec profile for this call */ - ast_format_cap_copy(p->peercaps, newpeercapability); /* The other side's capability in latest offer */ + ast_format_cap_remove_by_type(p->jointcaps, AST_MEDIA_TYPE_UNKNOWN); + ast_format_cap_append_from_cap(p->jointcaps, newjointcapability, AST_MEDIA_TYPE_UNKNOWN); /* Our joint codec profile for this call */ + ast_format_cap_remove_by_type(p->peercaps, AST_MEDIA_TYPE_UNKNOWN); + ast_format_cap_append_from_cap(p->peercaps, newpeercapability, AST_MEDIA_TYPE_UNKNOWN); /* The other side's capability in latest offer */ p->jointnoncodeccapability = newnoncodeccapability; /* DTMF capabilities */ /* respond with single most preferred joint codec, limiting the other side's choice */ if (ast_test_flag(&p->flags[1], SIP_PAGE2_PREFERRED_CODEC)) { - ast_codec_choose(&p->prefs, p->jointcaps, 1, &tmp_fmt); - ast_format_cap_set(p->jointcaps, &tmp_fmt); + unsigned int framing; + + tmp_fmt = ast_format_cap_get_format(p->jointcaps, 0); + framing = ast_format_cap_get_format_framing(p->jointcaps, tmp_fmt); + ast_format_cap_remove_by_type(p->jointcaps, AST_MEDIA_TYPE_UNKNOWN); + ast_format_cap_append(p->jointcaps, tmp_fmt, framing); + ao2_ref(tmp_fmt, -1); } } @@ -10813,7 +10870,7 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action ast_verbose("Peer T.140 RTP is at port %s\n", ast_sockaddr_stringify(tsa)); } - if (ast_format_cap_iscompatible(p->jointcaps, ast_format_set(&tmp_fmt, AST_FORMAT_T140RED, 0))) { + if (ast_format_cap_iscompatible_format(p->jointcaps, ast_format_t140_red) != AST_FORMAT_CMP_NOT_EQUAL) { p->red = 1; ast_rtp_red_init(p->trtp, 300, red_data_pt, 2); } else { @@ -10890,7 +10947,7 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action } /* Ok, we're going with this offer */ - ast_debug(2, "We're settling with these formats: %s\n", ast_getformatname_multiple(buf, SIPBUFSIZE, p->jointcaps)); + ast_debug(2, "We're settling with these formats: %s\n", ast_format_cap_get_names(p->jointcaps, &codec_buf)); if (!p->owner) { /* There's no open channel owning us so we can return here. For a re-invite or so, we proceed */ res = 0; @@ -10898,19 +10955,30 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action } ast_debug(4, "We have an owner, now see if we need to change this call\n"); - if (ast_format_cap_has_type(p->jointcaps, AST_FORMAT_TYPE_AUDIO)) { + if (ast_format_cap_has_type(p->jointcaps, AST_MEDIA_TYPE_AUDIO)) { + struct ast_format_cap *caps; + unsigned int framing; + if (debug) { - char s1[SIPBUFSIZE], s2[SIPBUFSIZE]; + struct ast_str *cap_buf = ast_str_alloca(64); + struct ast_str *joint_buf = ast_str_alloca(64); + ast_debug(1, "Setting native formats after processing SDP. peer joint formats %s, old nativeformats %s\n", - ast_getformatname_multiple(s1, SIPBUFSIZE, p->jointcaps), - ast_getformatname_multiple(s2, SIPBUFSIZE, ast_channel_nativeformats(p->owner))); + ast_format_cap_get_names(p->jointcaps, &joint_buf), + ast_format_cap_get_names(ast_channel_nativeformats(p->owner), &cap_buf)); } - ast_codec_choose(&p->prefs, p->jointcaps, 1, &tmp_fmt); - - ast_format_cap_set(ast_channel_nativeformats(p->owner), &tmp_fmt); - ast_format_cap_joint_append(p->caps, vpeercapability, ast_channel_nativeformats(p->owner)); - ast_format_cap_joint_append(p->caps, tpeercapability, ast_channel_nativeformats(p->owner)); + caps = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT); + if (caps) { + tmp_fmt = ast_format_cap_get_format(p->jointcaps, 0); + framing = ast_format_cap_get_format_framing(p->jointcaps, tmp_fmt); + ast_format_cap_append(caps, tmp_fmt, framing); + ast_format_cap_append_from_cap(caps, vpeercapability, AST_MEDIA_TYPE_VIDEO); + ast_format_cap_append_from_cap(caps, tpeercapability, AST_MEDIA_TYPE_TEXT); + ast_channel_nativeformats_set(p->owner, caps); + ao2_ref(caps, -1); + ao2_ref(tmp_fmt, -1); + } ast_set_read_format(p->owner, ast_channel_readformat(p->owner)); ast_set_write_format(p->owner, ast_channel_writeformat(p->owner)); @@ -10942,11 +11010,11 @@ process_sdp_cleanup: ast_rtp_codecs_payloads_destroy(&newtextrtp); ast_rtp_codecs_payloads_destroy(&newvideortp); ast_rtp_codecs_payloads_destroy(&newaudiortp); - ast_format_cap_destroy(peercapability); - ast_format_cap_destroy(vpeercapability); - ast_format_cap_destroy(tpeercapability); - ast_format_cap_destroy(newjointcapability); - ast_format_cap_destroy(newpeercapability); + ao2_cleanup(peercapability); + ao2_cleanup(vpeercapability); + ao2_cleanup(tpeercapability); + ao2_cleanup(newjointcapability); + ao2_cleanup(newpeercapability); return res; } @@ -11203,17 +11271,10 @@ static int process_sdp_a_audio(const char *a, struct sip_pvt *p, struct ast_rtp_ ast_debug(1, "Can't read framing from SDP: %s\n", a); } } + if (framing && p->autoframing) { - struct ast_codec_pref *pref = &ast_rtp_instance_get_codecs(p->rtp)->pref; - int codec_n; - for (codec_n = 0; codec_n < AST_RTP_MAX_PT; codec_n++) { - struct ast_rtp_payload_type format = ast_rtp_codecs_payload_lookup(ast_rtp_instance_get_codecs(p->rtp), codec_n); - if (!format.asterisk_format) /* non-codec or not found */ - continue; - ast_debug(1, "Setting framing for %s to %ld\n", ast_getformatname(&format.format), framing); - ast_codec_pref_setsize(pref, &format.format, framing); - } - ast_rtp_codecs_packetization_set(ast_rtp_instance_get_codecs(p->rtp), p->rtp, pref); + ast_debug(1, "Setting framing to %ld\n", framing); + ast_rtp_codecs_set_framing(ast_rtp_instance_get_codecs(p->rtp), framing); } found = TRUE; } else if (sscanf(a, "rtpmap: %30u %127[^/]/%30u", &codec, mimeSubtype, &sample_rate) == 3) { @@ -11240,15 +11301,19 @@ static int process_sdp_a_audio(const char *a, struct sip_pvt *p, struct ast_rtp_ if ((format = ast_rtp_codecs_get_payload_format(newaudiortp, codec))) { unsigned int bit_rate; + struct ast_format *format_parsed; - if (!ast_format_sdp_parse(format, fmtp_string)) { + format_parsed = ast_format_parse_sdp_fmtp(format, fmtp_string); + if (format_parsed) { + ast_rtp_codecs_payload_replace_format(newaudiortp, codec, format_parsed); + ao2_replace(format, format_parsed); + ao2_ref(format_parsed, -1); found = TRUE; } else { ast_rtp_codecs_payloads_unset(newaudiortp, NULL, codec); } - switch ((int) format->id) { - case AST_FORMAT_SIREN7: + if (ast_format_cmp(format, ast_format_siren7) == AST_FORMAT_CMP_EQUAL) { if (sscanf(fmtp_string, "bitrate=%30u", &bit_rate) == 1) { if (bit_rate != 32000) { ast_log(LOG_WARNING, "Got Siren7 offer at %u bps, but only 32000 bps supported; ignoring.\n", bit_rate); @@ -11257,8 +11322,7 @@ static int process_sdp_a_audio(const char *a, struct sip_pvt *p, struct ast_rtp_ found = TRUE; } } - break; - case AST_FORMAT_SIREN14: + } else if (ast_format_cmp(format, ast_format_siren14) == AST_FORMAT_CMP_EQUAL) { if (sscanf(fmtp_string, "bitrate=%30u", &bit_rate) == 1) { if (bit_rate != 48000) { ast_log(LOG_WARNING, "Got Siren14 offer at %u bps, but only 48000 bps supported; ignoring.\n", bit_rate); @@ -11267,8 +11331,7 @@ static int process_sdp_a_audio(const char *a, struct sip_pvt *p, struct ast_rtp_ found = TRUE; } } - break; - case AST_FORMAT_G719: + } else if (ast_format_cmp(format, ast_format_g719) == AST_FORMAT_CMP_EQUAL) { if (sscanf(fmtp_string, "bitrate=%30u", &bit_rate) == 1) { if (bit_rate != 64000) { ast_log(LOG_WARNING, "Got G.719 offer at %u bps, but only 64000 bps supported; ignoring.\n", bit_rate); @@ -11277,8 +11340,8 @@ static int process_sdp_a_audio(const char *a, struct sip_pvt *p, struct ast_rtp_ found = TRUE; } } - break; } + ao2_ref(format, -1); } } @@ -11320,11 +11383,19 @@ static int process_sdp_a_video(const char *a, struct sip_pvt *p, struct ast_rtp_ struct ast_format *format; if ((format = ast_rtp_codecs_get_payload_format(newvideortp, codec))) { - if (!ast_format_sdp_parse(format, fmtp_string)) { + struct ast_format *format_parsed; + + format_parsed = ast_format_parse_sdp_fmtp(format, fmtp_string); + + if (format_parsed) { + ast_rtp_codecs_payload_replace_format(newvideortp, codec, format_parsed); + ao2_replace(format, format_parsed); + ao2_ref(format_parsed, -1); found = TRUE; } else { ast_rtp_codecs_payloads_unset(newvideortp, NULL, codec); } + ao2_ref(format, -1); } } @@ -12890,12 +12961,11 @@ static void add_codec_to_sdp(const struct sip_pvt *p, int *max_packet_size) { int rtp_code; - struct ast_format_list fmt; const char *mime; - unsigned int rate; + unsigned int rate, framing; if (debug) - ast_verbose("Adding codec %u (%s) to SDP\n", format->id, ast_getformatname(format)); + ast_verbose("Adding codec %s to SDP\n", ast_format_get_name(format)); if (((rtp_code = ast_rtp_codecs_payload_code(ast_rtp_instance_get_codecs(p->rtp), 1, format, 0)) == -1) || !(mime = ast_rtp_lookup_mime_subtype2(1, format, 0, ast_test_flag(&p->flags[0], SIP_G726_NONSTANDARD) ? AST_RTP_OPT_G726_NONSTANDARD : 0)) || @@ -12903,63 +12973,54 @@ static void add_codec_to_sdp(const struct sip_pvt *p, return; } - if (p->rtp) { - struct ast_codec_pref *pref = &ast_rtp_instance_get_codecs(p->rtp)->pref; - fmt = ast_codec_pref_getsize(pref, format); - } else /* I don't see how you couldn't have p->rtp, but good to check for and error out if not there like earlier code */ - return; ast_str_append(m_buf, 0, " %d", rtp_code); /* Opus mandates 2 channels in rtpmap */ - if ((int)format->id == AST_FORMAT_OPUS) { + if (ast_format_cmp(format, ast_format_opus) == AST_FORMAT_CMP_EQUAL) { ast_str_append(a_buf, 0, "a=rtpmap:%d %s/%u/2\r\n", rtp_code, mime, rate); } else { ast_str_append(a_buf, 0, "a=rtpmap:%d %s/%u\r\n", rtp_code, mime, rate); } - ast_format_sdp_generate(format, rtp_code, a_buf); + ast_format_generate_sdp_fmtp(format, rtp_code, a_buf); + + framing = ast_format_cap_get_format_framing(p->caps, format); - switch ((int) format->id) { - case AST_FORMAT_G729A: + if (ast_format_cmp(format, ast_format_g729) == AST_FORMAT_CMP_EQUAL) { /* Indicate that we don't support VAD (G.729 annex B) */ ast_str_append(a_buf, 0, "a=fmtp:%d annexb=no\r\n", rtp_code); - break; - case AST_FORMAT_G723_1: + } else if (ast_format_cmp(format, ast_format_g723) == AST_FORMAT_CMP_EQUAL) { /* Indicate that we don't support VAD (G.723.1 annex A) */ ast_str_append(a_buf, 0, "a=fmtp:%d annexa=no\r\n", rtp_code); - break; - case AST_FORMAT_ILBC: + } else if (ast_format_cmp(format, ast_format_ilbc) == AST_FORMAT_CMP_EQUAL) { /* Add information about us using only 20/30 ms packetization */ - ast_str_append(a_buf, 0, "a=fmtp:%d mode=%d\r\n", rtp_code, fmt.cur_ms); - break; - case AST_FORMAT_SIREN7: + ast_str_append(a_buf, 0, "a=fmtp:%d mode=%u\r\n", rtp_code, framing); + } else if (ast_format_cmp(format, ast_format_siren7) == AST_FORMAT_CMP_EQUAL) { /* Indicate that we only expect 32Kbps */ ast_str_append(a_buf, 0, "a=fmtp:%d bitrate=32000\r\n", rtp_code); - break; - case AST_FORMAT_SIREN14: + } else if (ast_format_cmp(format, ast_format_siren14) == AST_FORMAT_CMP_EQUAL) { /* Indicate that we only expect 48Kbps */ ast_str_append(a_buf, 0, "a=fmtp:%d bitrate=48000\r\n", rtp_code); - break; - case AST_FORMAT_G719: + } else if (ast_format_cmp(format, ast_format_g719) == AST_FORMAT_CMP_EQUAL) { /* Indicate that we only expect 64Kbps */ ast_str_append(a_buf, 0, "a=fmtp:%d bitrate=64000\r\n", rtp_code); - break; } - if (max_packet_size && fmt.max_ms && (fmt.max_ms < *max_packet_size)) { - *max_packet_size = fmt.max_ms; + if (max_packet_size && ast_format_get_maximum_ms(format) && + (ast_format_get_maximum_ms(format) < *max_packet_size)) { + *max_packet_size = ast_format_get_maximum_ms(format); } - if (fmt.cur_ms && (fmt.cur_ms < *min_packet_size)) { - *min_packet_size = fmt.cur_ms; + if (framing && (framing < *min_packet_size)) { + *min_packet_size = framing; } /* Our first codec packetization processed cannot be zero */ - if ((*min_packet_size) == 0 && fmt.cur_ms) { - *min_packet_size = fmt.cur_ms; + if ((*min_packet_size) == 0 && framing) { + *min_packet_size = framing; } - if ((*max_packet_size) == 0 && fmt.max_ms) { - *max_packet_size = fmt.max_ms; + if ((*max_packet_size) == 0 && ast_format_get_maximum_ms(format)) { + *max_packet_size = ast_format_get_maximum_ms(format); } } @@ -12977,7 +13038,7 @@ static void add_vcodec_to_sdp(const struct sip_pvt *p, struct ast_format *format return; if (debug) - ast_verbose("Adding video codec %u (%s) to SDP\n", format->id, ast_getformatname(format)); + ast_verbose("Adding video codec %s to SDP\n", ast_format_get_name(format)); if (((rtp_code = ast_rtp_codecs_payload_code(ast_rtp_instance_get_codecs(p->vrtp), 1, format, 0)) == -1) || !(subtype = ast_rtp_lookup_mime_subtype2(1, format, 0, 0)) || @@ -12988,11 +13049,11 @@ static void add_vcodec_to_sdp(const struct sip_pvt *p, struct ast_format *format ast_str_append(m_buf, 0, " %d", rtp_code); ast_str_append(a_buf, 0, "a=rtpmap:%d %s/%u\r\n", rtp_code, subtype, rate); /* VP8: add RTCP FIR support */ - if ((int)format->id == AST_FORMAT_VP8) { + if (ast_format_cmp(format, ast_format_vp8) == AST_FORMAT_CMP_EQUAL) { ast_str_append(a_buf, 0, "a=rtcp-fb:* ccm fir\r\n"); } - ast_format_sdp_generate(format, rtp_code, a_buf); + ast_format_generate_sdp_fmtp(format, rtp_code, a_buf); } /*! \brief Add text codec offer to SDP offer/answer body in INVITE or 200 OK */ @@ -13006,7 +13067,7 @@ static void add_tcodec_to_sdp(const struct sip_pvt *p, struct ast_format *format return; if (debug) - ast_verbose("Adding text codec %u (%s) to SDP\n", format->id, ast_getformatname(format)); + ast_verbose("Adding text codec %s to SDP\n", ast_format_get_name(format)); if ((rtp_code = ast_rtp_codecs_payload_code(ast_rtp_instance_get_codecs(p->trtp), 1, format, 0)) == -1) return; @@ -13017,9 +13078,8 @@ static void add_tcodec_to_sdp(const struct sip_pvt *p, struct ast_format *format ast_rtp_lookup_sample_rate2(1, format, 0)); /* Add fmtp code here */ - if (format->id == AST_FORMAT_T140RED) { - struct ast_format tmp_fmt; - int t140code = ast_rtp_codecs_payload_code(ast_rtp_instance_get_codecs(p->trtp), 1, ast_format_set(&tmp_fmt, AST_FORMAT_T140, 0), 0); + if (ast_format_cmp(format, ast_format_t140_red) == AST_FORMAT_CMP_EQUAL) { + int t140code = ast_rtp_codecs_payload_code(ast_rtp_instance_get_codecs(p->trtp), 1, ast_format_t140, 0); ast_str_append(a_buf, 0, "a=fmtp:%d %d/%d/%d\r\n", rtp_code, t140code, t140code, @@ -13194,8 +13254,8 @@ static char *crypto_get_attrib(struct ast_sdp_srtp *srtp, int dtls_enabled, int */ static enum sip_result add_sdp(struct sip_request *resp, struct sip_pvt *p, int oldsdp, int add_audio, int add_t38) { - struct ast_format_cap *alreadysent = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_NOLOCK); - struct ast_format_cap *tmpcap = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_NOLOCK); + struct ast_format_cap *alreadysent = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT); + struct ast_format_cap *tmpcap = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT); int res = AST_SUCCESS; int doing_directmedia = FALSE; struct ast_sockaddr addr = { {0,} }; @@ -13229,7 +13289,7 @@ static enum sip_result add_sdp(struct sip_request *resp, struct sip_pvt *p, int RAII_VAR(char *, t_a_crypto, NULL, ast_free); int x; - struct ast_format tmp_fmt; + struct ast_format *tmp_fmt; int needaudio = FALSE; int needvideo = FALSE; int needtext = FALSE; @@ -13239,8 +13299,7 @@ static enum sip_result add_sdp(struct sip_request *resp, struct sip_pvt *p, int int min_video_packet_size = 0; int min_text_packet_size = 0; - char codecbuf[SIPBUFSIZE]; - char buf[SIPBUFSIZE]; + struct ast_str *codec_buf = ast_str_alloca(64); /* Set the SDP session name */ snprintf(subject, sizeof(subject), "s=%s\r\n", ast_strlen_zero(global_sdpsession) ? "-" : global_sdpsession); @@ -13268,11 +13327,24 @@ static enum sip_result add_sdp(struct sip_request *resp, struct sip_pvt *p, int } if (add_audio) { - doing_directmedia = (!ast_sockaddr_isnull(&p->redirip) && !(ast_format_cap_is_empty(p->redircaps))) ? TRUE : FALSE; + doing_directmedia = (!ast_sockaddr_isnull(&p->redirip) && (ast_format_cap_count(p->redircaps))) ? TRUE : FALSE; + + if (doing_directmedia) { + ast_format_cap_get_compatible(p->jointcaps, p->redircaps, tmpcap); + ast_debug(1, "** Our native-bridge filtered capablity: %s\n", ast_format_cap_get_names(tmpcap, &codec_buf)); + } else { + ast_format_cap_append_from_cap(tmpcap, p->jointcaps, AST_MEDIA_TYPE_UNKNOWN); + } + + /* Check if we need audio */ + if (ast_format_cap_has_type(tmpcap, AST_MEDIA_TYPE_AUDIO) + || ast_format_cap_has_type(p->caps, AST_MEDIA_TYPE_AUDIO)) { + needaudio = TRUE; + } + /* Check if we need video in this call */ - if ((ast_format_cap_has_type(p->jointcaps, AST_FORMAT_TYPE_VIDEO)) && !p->novideo) { - ast_format_cap_joint_copy(p->jointcaps, p->redircaps, tmpcap); - if (doing_directmedia && !ast_format_cap_has_type(tmpcap, AST_FORMAT_TYPE_VIDEO)) { + if ((ast_format_cap_has_type(tmpcap, AST_MEDIA_TYPE_VIDEO)) && !p->novideo) { + if (doing_directmedia && !ast_format_cap_has_type(tmpcap, AST_MEDIA_TYPE_VIDEO)) { ast_debug(2, "This call needs video offers, but caller probably did not offer it!\n"); } else if (p->vrtp) { needvideo = TRUE; @@ -13281,8 +13353,9 @@ static enum sip_result add_sdp(struct sip_request *resp, struct sip_pvt *p, int ast_debug(2, "This call needs video offers, but there's no video support enabled!\n"); } } + /* Check if we need text in this call */ - if ((ast_format_cap_has_type(p->jointcaps, AST_FORMAT_TYPE_TEXT)) && !p->notext) { + if ((ast_format_cap_has_type(p->jointcaps, AST_MEDIA_TYPE_TEXT)) && !p->notext) { if (sipdebug_text) ast_verbose("We think we can do text\n"); if (p->trtp) { @@ -13295,6 +13368,12 @@ static enum sip_result add_sdp(struct sip_request *resp, struct sip_pvt *p, int ast_debug(2, "This call needs text offers, but there's no text support enabled ! \n"); } } + + /* XXX note, Video and Text are negated - 'true' means 'no' */ + ast_debug(1, "** Our capability: %s Video flag: %s Text flag: %s\n", + ast_format_cap_get_names(tmpcap, &codec_buf), + p->novideo ? "True" : "False", p->notext ? "True" : "False"); + ast_debug(1, "** Our prefcodec: %s \n", ast_format_cap_get_names(p->prefcaps, &codec_buf)); } get_our_media_address(p, needvideo, needtext, &addr, &vaddr, &taddr, &dest, &vdest, &tdest); @@ -13322,22 +13401,6 @@ static enum sip_result add_sdp(struct sip_request *resp, struct sip_pvt *p, int hold = "a=sendrecv\r\n"; } - ast_format_cap_copy(tmpcap, p->jointcaps); - - /* XXX note, Video and Text are negated - 'true' means 'no' */ - ast_debug(1, "** Our capability: %s Video flag: %s Text flag: %s\n", ast_getformatname_multiple(codecbuf, sizeof(codecbuf), tmpcap), - p->novideo ? "True" : "False", p->notext ? "True" : "False"); - ast_debug(1, "** Our prefcodec: %s \n", ast_getformatname_multiple(codecbuf, sizeof(codecbuf), p->prefcaps)); - - if (doing_directmedia) { - ast_format_cap_joint_copy(p->jointcaps, p->redircaps, tmpcap); - ast_debug(1, "** Our native-bridge filtered capablity: %s\n", ast_getformatname_multiple(codecbuf, sizeof(codecbuf), tmpcap)); - } - - /* Check if we need audio */ - if (ast_format_cap_has_type(tmpcap, AST_FORMAT_TYPE_AUDIO)) - needaudio = TRUE; - if (debug) { ast_verbose("Audio is at %s\n", ast_sockaddr_stringify_port(&addr)); } @@ -13406,8 +13469,8 @@ static enum sip_result add_sdp(struct sip_request *resp, struct sip_pvt *p, int /* Now, start adding audio codecs. These are added in this order: - First what was requested by the calling channel + - Then our mutually shared capabilities, determined previous in tmpcap - Then preferences in order from sip.conf device config for this peer/user - - Then other codecs in capabilities, including video */ @@ -13415,57 +13478,62 @@ static enum sip_result add_sdp(struct sip_request *resp, struct sip_pvt *p, int * configured codecs. */ if (!ast_test_flag(&p->flags[2], SIP_PAGE3_IGNORE_PREFCAPS)) { - ast_format_cap_iter_start(p->prefcaps); - while (!(ast_format_cap_iter_next(p->prefcaps, &tmp_fmt))) { - if (AST_FORMAT_GET_TYPE(tmp_fmt.id) != AST_FORMAT_TYPE_AUDIO || - !ast_format_cap_iscompatible(tmpcap, &tmp_fmt)) { + for (x = 0; x < ast_format_cap_count(p->prefcaps); x++) { + tmp_fmt = ast_format_cap_get_format(p->prefcaps, x); + + if ((ast_format_get_type(tmp_fmt) != AST_MEDIA_TYPE_AUDIO) || + (ast_format_cap_iscompatible_format(tmpcap, tmp_fmt) == AST_FORMAT_CMP_NOT_EQUAL)) { + ao2_ref(tmp_fmt, -1); continue; } - add_codec_to_sdp(p, &tmp_fmt, &m_audio, &a_audio, debug, &min_audio_packet_size, &max_audio_packet_size); - ast_format_cap_add(alreadysent, &tmp_fmt); + + add_codec_to_sdp(p, tmp_fmt, &m_audio, &a_audio, debug, &min_audio_packet_size, &max_audio_packet_size); + ast_format_cap_append(alreadysent, tmp_fmt, 0); + ao2_ref(tmp_fmt, -1); } - ast_format_cap_iter_end(p->prefcaps); } - /* Start by sending our preferred audio/video codecs */ - for (x = 0; x < AST_CODEC_PREF_SIZE; x++) { - struct ast_format pref; - - if (!(ast_codec_pref_index(&p->prefs, x, &pref))) - break; - - if (!ast_format_cap_get_compatible_format(tmpcap, &pref, &tmp_fmt)) - continue; + /* Now send any other common codecs */ + for (x = 0; x < ast_format_cap_count(tmpcap); x++) { + tmp_fmt = ast_format_cap_get_format(tmpcap, x); - if (ast_format_cap_iscompatible(alreadysent, &tmp_fmt)) + if (ast_format_cap_iscompatible_format(alreadysent, tmp_fmt) != AST_FORMAT_CMP_NOT_EQUAL) { + ao2_ref(tmp_fmt, -1); continue; + } - if (AST_FORMAT_GET_TYPE(tmp_fmt.id) == AST_FORMAT_TYPE_AUDIO) { - add_codec_to_sdp(p, &tmp_fmt, &m_audio, &a_audio, debug, &min_audio_packet_size, &max_audio_packet_size); - } else if (needvideo && (AST_FORMAT_GET_TYPE(tmp_fmt.id) == AST_FORMAT_TYPE_VIDEO)) { - add_vcodec_to_sdp(p, &tmp_fmt, &m_video, &a_video, debug, &min_video_packet_size); - } else if (needtext && (AST_FORMAT_GET_TYPE(tmp_fmt.id) == AST_FORMAT_TYPE_TEXT)) { - add_tcodec_to_sdp(p, &tmp_fmt, &m_text, &a_text, debug, &min_text_packet_size); + if (ast_format_get_type(tmp_fmt) == AST_MEDIA_TYPE_AUDIO) { + add_codec_to_sdp(p, tmp_fmt, &m_audio, &a_audio, debug, &min_audio_packet_size, &max_audio_packet_size); + } else if (needvideo && ast_format_get_type(tmp_fmt) == AST_MEDIA_TYPE_VIDEO) { + add_vcodec_to_sdp(p, tmp_fmt, &m_video, &a_video, debug, &min_video_packet_size); + } else if (needtext && ast_format_get_type(tmp_fmt) == AST_MEDIA_TYPE_TEXT) { + add_tcodec_to_sdp(p, tmp_fmt, &m_text, &a_text, debug, &min_text_packet_size); } - ast_format_cap_add(alreadysent, &tmp_fmt); + ast_format_cap_append(alreadysent, tmp_fmt, 0); + ao2_ref(tmp_fmt, -1); } - /* Now send any other common audio and video codecs, and non-codec formats: */ - ast_format_cap_iter_start(tmpcap); - while (!(ast_format_cap_iter_next(tmpcap, &tmp_fmt))) { - if (ast_format_cap_iscompatible(alreadysent, &tmp_fmt)) + /* Finally our remaining audio/video codecs */ + for (x = 0; x < ast_format_cap_count(p->caps); x++) { + tmp_fmt = ast_format_cap_get_format(p->caps, x); + + if (ast_format_cap_iscompatible_format(alreadysent, tmp_fmt) != AST_FORMAT_CMP_NOT_EQUAL) { + ao2_ref(tmp_fmt, -1); continue; + } - if (AST_FORMAT_GET_TYPE(tmp_fmt.id) == AST_FORMAT_TYPE_AUDIO) { - add_codec_to_sdp(p, &tmp_fmt, &m_audio, &a_audio, debug, &min_audio_packet_size, &max_audio_packet_size); - } else if (needvideo && (AST_FORMAT_GET_TYPE(tmp_fmt.id) == AST_FORMAT_TYPE_VIDEO)) { - add_vcodec_to_sdp(p, &tmp_fmt, &m_video, &a_video, debug, &min_video_packet_size); - } else if (needtext && (AST_FORMAT_GET_TYPE(tmp_fmt.id) == AST_FORMAT_TYPE_TEXT)) { - add_tcodec_to_sdp(p, &tmp_fmt, &m_text, &a_text, debug, &min_text_packet_size); + if (ast_format_get_type(tmp_fmt) == AST_MEDIA_TYPE_AUDIO) { + add_codec_to_sdp(p, tmp_fmt, &m_audio, &a_audio, debug, &min_audio_packet_size, &max_audio_packet_size); + } else if (needvideo && ast_format_get_type(tmp_fmt) == AST_MEDIA_TYPE_VIDEO) { + add_vcodec_to_sdp(p, tmp_fmt, &m_video, &a_video, debug, &min_video_packet_size); + } else if (needtext && ast_format_get_type(tmp_fmt) == AST_MEDIA_TYPE_TEXT) { + add_tcodec_to_sdp(p, tmp_fmt, &m_text, &a_text, debug, &min_text_packet_size); } + + ast_format_cap_append(alreadysent, tmp_fmt, 0); + ao2_ref(tmp_fmt, -1); } - ast_format_cap_iter_end(tmpcap); /* Now add DTMF RFC2833 telephony-event as a codec */ for (x = 1LL; x <= AST_RTP_MAX; x <<= 1) { @@ -13677,14 +13745,15 @@ static enum sip_result add_sdp(struct sip_request *resp, struct sip_pvt *p, int ao2_t_link(dialogs_rtpcheck, p, "link pvt into dialogs_rtpcheck container"); ao2_unlock(dialogs_rtpcheck); - ast_debug(3, "Done building SDP. Settling with this capability: %s\n", ast_getformatname_multiple(buf, SIPBUFSIZE, tmpcap)); + ast_debug(3, "Done building SDP. Settling with this capability: %s\n", + ast_format_cap_get_names(tmpcap, &codec_buf)); add_sdp_cleanup: ast_free(a_text); ast_free(a_video); ast_free(a_audio); - alreadysent = ast_format_cap_destroy(alreadysent); - tmpcap = ast_format_cap_destroy(tmpcap); + ao2_cleanup(alreadysent); + ao2_cleanup(tmpcap); return res; } @@ -13792,7 +13861,7 @@ static int transmit_response_with_sdp(struct sip_pvt *p, const char *msg, const if (p->rtp) { if (!p->autoframing && !ast_test_flag(&p->flags[0], SIP_OUTGOING)) { ast_debug(1, "Setting framing from config on incoming call\n"); - ast_rtp_codecs_packetization_set(ast_rtp_instance_get_codecs(p->rtp), p->rtp, &p->prefs); + ast_rtp_codecs_set_framing(ast_rtp_instance_get_codecs(p->rtp), ast_format_cap_get_framing(p->caps)); } ast_rtp_instance_activate(p->rtp); try_suggested_sip_codec(p); @@ -18475,10 +18544,9 @@ static enum check_auth_result check_peer_ok(struct sip_pvt *p, char *of, peer->name, of, ast_sockaddr_stringify(&p->recv)); } - /* XXX what about p->prefs = peer->prefs; ? */ /* Set Frame packetization */ if (p->rtp) { - ast_rtp_codecs_packetization_set(ast_rtp_instance_get_codecs(p->rtp), p->rtp, &peer->prefs); + ast_rtp_codecs_set_framing(ast_rtp_instance_get_codecs(p->rtp), ast_format_cap_get_framing(peer->caps)); p->autoframing = peer->autoframing; } @@ -18595,20 +18663,22 @@ static enum check_auth_result check_peer_ok(struct sip_pvt *p, char *of, p->named_callgroups = ast_ref_namedgroups(peer->named_callgroups); ast_unref_namedgroups(p->named_pickupgroups); p->named_pickupgroups = ast_ref_namedgroups(peer->named_pickupgroups); - ast_format_cap_copy(p->caps, peer->caps); - ast_format_cap_copy(p->jointcaps, peer->caps); - p->prefs = peer->prefs; + ast_format_cap_remove_by_type(p->caps, AST_MEDIA_TYPE_UNKNOWN); + ast_format_cap_append_from_cap(p->caps, peer->caps, AST_MEDIA_TYPE_UNKNOWN); + ast_format_cap_remove_by_type(p->jointcaps, AST_MEDIA_TYPE_UNKNOWN); + ast_format_cap_append_from_cap(p->jointcaps, peer->caps, AST_MEDIA_TYPE_UNKNOWN); ast_copy_string(p->zone, peer->zone, sizeof(p->zone)); if (peer->maxforwards > 0) { p->maxforwards = peer->maxforwards; } - if (!(ast_format_cap_is_empty(p->peercaps))) { - struct ast_format_cap *tmp = ast_format_cap_joint(p->jointcaps, p->peercaps); - struct ast_format_cap *tmp2; - if (tmp) { - tmp2 = p->jointcaps; - p->jointcaps = tmp; - ast_format_cap_destroy(tmp2); + if (ast_format_cap_count(p->peercaps)) { + struct ast_format_cap *joint; + + joint = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT); + if (joint) { + ast_format_cap_get_compatible(p->jointcaps, p->peercaps, joint); + ao2_ref(p->jointcaps, -1); + p->jointcaps = joint; } } p->maxcallbitrate = peer->maxcallbitrate; @@ -18623,7 +18693,7 @@ static enum check_auth_result check_peer_ok(struct sip_pvt *p, char *of, p->rtpkeepalive = peer->rtpkeepalive; if (!dialog_initialize_rtp(p)) { if (p->rtp) { - ast_rtp_codecs_packetization_set(ast_rtp_instance_get_codecs(p->rtp), p->rtp, &peer->prefs); + ast_rtp_codecs_set_framing(ast_rtp_instance_get_codecs(p->rtp), ast_format_cap_get_framing(peer->caps)); p->autoframing = peer->autoframing; } } else { @@ -19959,25 +20029,6 @@ static char *sip_prune_realtime(struct ast_cli_entry *e, int cmd, struct ast_cli return CLI_SUCCESS; } -/*! \brief Print codec list from preference to CLI/manager */ -static void print_codec_to_cli(int fd, struct ast_codec_pref *pref) -{ - int x; - struct ast_format codec; - - for(x = 0; x < AST_CODEC_PREF_SIZE; x++) { - if (!(ast_codec_pref_index(pref, x, &codec))) { - break; - } - ast_cli(fd, "%s", ast_getformatname(&codec)); - ast_cli(fd, ":%d", pref->framing[x]); - if (x < 31 && ast_codec_pref_index(pref, x + 1, &codec)) - ast_cli(fd, ","); - } - if (!x) - ast_cli(fd, "none"); -} - /*! \brief Print domain mode to cli */ static const char *domain_mode_to_text(const enum domain_mode mode) { @@ -20262,11 +20313,9 @@ static char *_sip_show_peer(int type, int fd, struct mansession *s, const struct char status[30] = ""; char cbuf[256]; struct sip_peer *peer; - char codec_buf[512]; - struct ast_codec_pref *pref; + struct ast_str *codec_buf = ast_str_alloca(64); struct ast_variable *v; int x = 0, load_realtime; - struct ast_format codec; int realtimepeers; realtimepeers = ast_check_realtime("sippeers"); @@ -20415,12 +20464,7 @@ static char *_sip_show_peer(int type, int fd, struct mansession *s, const struct ast_cli(fd, "(none)"); ast_cli(fd, "\n"); - ast_cli(fd, " Codecs : "); - ast_getformatname_multiple(codec_buf, sizeof(codec_buf) -1, peer->caps); - ast_cli(fd, "%s\n", codec_buf); - ast_cli(fd, " Codec Order : ("); - print_codec_to_cli(fd, &peer->prefs); - ast_cli(fd, ")\n"); + ast_cli(fd, " Codecs : %s\n", ast_format_cap_get_names(peer->caps, &codec_buf)); ast_cli(fd, " Auto-Framing : %s\n", AST_CLI_YESNO(peer->autoframing)); ast_cli(fd, " Status : "); @@ -20523,21 +20567,7 @@ static char *_sip_show_peer(int type, int fd, struct mansession *s, const struct astman_append(s, "Default-Username: %s\r\n", peer->username); if (!ast_strlen_zero(sip_cfg.regcontext)) astman_append(s, "RegExtension: %s\r\n", peer->regexten); - astman_append(s, "Codecs: "); - ast_getformatname_multiple(codec_buf, sizeof(codec_buf) -1, peer->caps); - astman_append(s, "%s\r\n", codec_buf); - astman_append(s, "CodecOrder: "); - pref = &peer->prefs; - for(x = 0; x < AST_CODEC_PREF_SIZE ; x++) { - if (!(ast_codec_pref_index(pref, x, &codec))) { - break; - } - astman_append(s, "%s", ast_getformatname(&codec)); - if ((x < (AST_CODEC_PREF_SIZE - 1)) && ast_codec_pref_index(pref, x+1, &codec)) - astman_append(s, ","); - } - - astman_append(s, "\r\n"); + astman_append(s, "Codecs: %s\r\n", ast_format_cap_get_names(peer->caps, &codec_buf)); astman_append(s, "Status: "); peer_status(peer, status, sizeof(status)); astman_append(s, "%s\r\n", status); @@ -20660,10 +20690,6 @@ static char *sip_show_user(struct ast_cli_entry *e, int cmd, struct ast_cli_args ast_cli(a->fd, " Sess-Min-SE : %d secs\n", user->stimer.st_min_se); ast_cli(a->fd, " RTP Engine : %s\n", user->engine); - ast_cli(a->fd, " Codec Order : ("); - print_codec_to_cli(a->fd, &user->prefs); - ast_cli(a->fd, ")\n"); - ast_cli(a->fd, " Auto-Framing: %s \n", AST_CLI_YESNO(user->autoframing)); if (user->chanvars) { ast_cli(a->fd, " Variables :\n"); @@ -20916,7 +20942,7 @@ static char *sip_show_settings(struct ast_cli_entry *e, int cmd, struct ast_cli_ { int realtimepeers; int realtimeregs; - char codec_buf[SIPBUFSIZE]; + struct ast_str *codec_buf = ast_str_alloca(64); const char *msg; /* temporary msg pointer */ struct sip_auth_container *credentials; @@ -21069,12 +21095,7 @@ static char *sip_show_settings(struct ast_cli_entry *e, int cmd, struct ast_cli_ } ast_cli(a->fd, "\nGlobal Signalling Settings:\n"); ast_cli(a->fd, "---------------------------\n"); - ast_cli(a->fd, " Codecs: "); - ast_getformatname_multiple(codec_buf, sizeof(codec_buf) -1, sip_cfg.caps); - ast_cli(a->fd, "%s\n", codec_buf); - ast_cli(a->fd, " Codec Order: "); - print_codec_to_cli(a->fd, &default_prefs); - ast_cli(a->fd, "\n"); + ast_cli(a->fd, " Codecs: %s\n", ast_format_cap_get_names(sip_cfg.caps, &codec_buf)); ast_cli(a->fd, " Relax DTMF: %s\n", AST_CLI_YESNO(global_relaxdtmf)); ast_cli(a->fd, " RFC2833 Compensation: %s\n", AST_CLI_YESNO(ast_test_flag(&global_flags[1], SIP_PAGE2_RFC2833_COMPENSATE))); ast_cli(a->fd, " Symmetric RTP: %s\n", comedia_string(global_flags)); @@ -21233,12 +21254,12 @@ static int show_channels_cb(void *__cur, void *__arg, int flags) if (cur->subscribed == NONE && !arg->subscriptions) { /* set if SIP transfer in progress */ const char *referstatus = cur->refer ? referstatus2str(cur->refer->status) : ""; - char formatbuf[SIPBUFSIZE/2]; + struct ast_str *codec_buf = ast_str_alloca(64); ast_cli(arg->fd, FORMAT, ast_sockaddr_stringify_addr(dst), S_OR(cur->username, S_OR(cur->cid_num, "(None)")), cur->callid, - cur->owner ? ast_getformatname_multiple(formatbuf, sizeof(formatbuf), ast_channel_nativeformats(cur->owner)) : "(nothing)", + cur->owner ? ast_format_cap_get_names(ast_channel_nativeformats(cur->owner), &codec_buf) : "(nothing)", AST_CLI_YESNO(ast_test_flag(&cur->flags[1], SIP_PAGE2_CALL_ONHOLD)), cur->needdestroy ? "(d)" : "", cur->lastmsg , @@ -21479,7 +21500,8 @@ static char *sip_show_channel(struct ast_cli_entry *e, int cmd, struct ast_cli_a if (!strncasecmp(cur->callid, a->argv[3], len)) { struct ast_str *strbuf; - char formatbuf[SIPBUFSIZE/2]; + struct ast_str *codec_buf = ast_str_alloca(64); + ast_cli(a->fd, "\n"); if (cur->subscribed != NONE) { ast_cli(a->fd, " * Subscription (type: %s)\n", subscription_type2str(cur->subscribed)); @@ -21489,11 +21511,11 @@ static char *sip_show_channel(struct ast_cli_entry *e, int cmd, struct ast_cli_a ast_cli(a->fd, " Curr. trans. direction: %s\n", ast_test_flag(&cur->flags[0], SIP_OUTGOING) ? "Outgoing" : "Incoming"); ast_cli(a->fd, " Call-ID: %s\n", cur->callid); ast_cli(a->fd, " Owner channel ID: %s\n", cur->owner ? ast_channel_name(cur->owner) : "<none>"); - ast_cli(a->fd, " Our Codec Capability: %s\n", ast_getformatname_multiple(formatbuf, sizeof(formatbuf), cur->caps)); + ast_cli(a->fd, " Our Codec Capability: %s\n", ast_format_cap_get_names(cur->caps, &codec_buf)); ast_cli(a->fd, " Non-Codec Capability (DTMF): %d\n", cur->noncodeccapability); - ast_cli(a->fd, " Their Codec Capability: %s\n", ast_getformatname_multiple(formatbuf, sizeof(formatbuf), cur->peercaps)); - ast_cli(a->fd, " Joint Codec Capability: %s\n", ast_getformatname_multiple(formatbuf, sizeof(formatbuf), cur->jointcaps)); - ast_cli(a->fd, " Format: %s\n", cur->owner ? ast_getformatname_multiple(formatbuf, sizeof(formatbuf), ast_channel_nativeformats(cur->owner)) : "(nothing)" ); + ast_cli(a->fd, " Their Codec Capability: %s\n", ast_format_cap_get_names(cur->peercaps, &codec_buf)); + ast_cli(a->fd, " Joint Codec Capability: %s\n", ast_format_cap_get_names(cur->jointcaps, &codec_buf)); + ast_cli(a->fd, " Format: %s\n", cur->owner ? ast_format_cap_get_names(ast_channel_nativeformats(cur->owner), &codec_buf) : "(nothing)" ); ast_cli(a->fd, " T.38 support %s\n", AST_CLI_YESNO(cur->udptl != NULL)); ast_cli(a->fd, " Video support %s\n", AST_CLI_YESNO(cur->vrtp != NULL)); ast_cli(a->fd, " MaxCallBR: %d kbps\n", cur->maxcallbitrate); @@ -22437,7 +22459,9 @@ static int function_sippeer(struct ast_channel *chan, const char *cmd, char *dat } else if (!strcasecmp(colname, "callerid_num")) { ast_copy_string(buf, peer->cid_num, len); } else if (!strcasecmp(colname, "codecs")) { - ast_getformatname_multiple(buf, len -1, peer->caps); + struct ast_str *codec_buf = ast_str_alloca(64); + ast_format_cap_get_names(peer->caps, &codec_buf); + ast_copy_string(buf, ast_str_buffer(codec_buf), len); } else if (!strcasecmp(colname, "encryption")) { snprintf(buf, len, "%u", ast_test_flag(&peer->flags[1], SIP_PAGE2_USE_SRTP)); } else if (!strncasecmp(colname, "chanvar[", 8)) { @@ -22452,12 +22476,14 @@ static int function_sippeer(struct ast_channel *chan, const char *cmd, char *dat } } else if (!strncasecmp(colname, "codec[", 6)) { char *codecnum; - struct ast_format codec; + struct ast_format *codec; codecnum = colname + 6; /* move past the '[' */ codecnum = strsep(&codecnum, "]"); /* trim trailing ']' if any */ - if((ast_codec_pref_index(&peer->prefs, atoi(codecnum), &codec))) { - ast_copy_string(buf, ast_getformatname(&codec), len); + codec = ast_format_cap_get_format(peer->caps, atoi(codecnum)); + if (codec) { + ast_copy_string(buf, ast_format_get_name(codec), len); + ao2_ref(codec, -1); } else { buf[0] = '\0'; } @@ -25476,7 +25502,8 @@ static int handle_request_invite(struct sip_pvt *p, struct sip_request *req, str } ast_queue_control(p->owner, AST_CONTROL_SRCUPDATE); } else { - ast_format_cap_copy(p->jointcaps, p->caps); + ast_format_cap_remove_by_type(p->jointcaps, AST_MEDIA_TYPE_UNKNOWN); + ast_format_cap_append_from_cap(p->jointcaps, p->caps, AST_MEDIA_TYPE_UNKNOWN); ast_debug(1, "Hm.... No sdp for the moment\n"); /* Some devices signal they want to be put off hold by sending a re-invite *without* an SDP, which is supposed to mean "Go back to your state" @@ -25541,7 +25568,8 @@ static int handle_request_invite(struct sip_pvt *p, struct sip_request *req, str goto request_invite_cleanup; } } else { /* No SDP in invite, call control session */ - ast_format_cap_copy(p->jointcaps, p->caps); + ast_format_cap_remove_by_type(p->jointcaps, AST_MEDIA_TYPE_UNKNOWN); + ast_format_cap_append_from_cap(p->jointcaps, p->caps, AST_MEDIA_TYPE_UNKNOWN); ast_debug(2, "No SDP in Invite, third party call control\n"); } @@ -25855,7 +25883,7 @@ static int handle_request_invite(struct sip_pvt *p, struct sip_request *req, str if (p && (p->autokillid == -1)) { const char *msg; - if ((ast_format_cap_is_empty(p->jointcaps))) + if ((!ast_format_cap_count(p->jointcaps))) msg = "488 Not Acceptable Here (codec error)"; else { ast_log(LOG_NOTICE, "Unable to create/find SIP channel for this INVITE\n"); @@ -29539,7 +29567,8 @@ static struct ast_channel *sip_request_call(const char *type, struct ast_format_ struct ast_channel *tmpc = NULL; char *ext = NULL, *host; char tmp[256]; - char tmp2[256]; + struct ast_str *codec_buf = ast_str_alloca(64); + struct ast_str *cap_buf = ast_str_alloca(64); char *dnid; char *secret = NULL; char *md5secret = NULL; @@ -29562,14 +29591,14 @@ static struct ast_channel *sip_request_call(const char *type, struct ast_format_ * configured from sip.conf, and sip_tech.capabilities, which is * hardwired to all audio formats. */ - if (!(ast_format_cap_has_type(cap, AST_FORMAT_TYPE_AUDIO))) { + if (!(ast_format_cap_has_type(cap, AST_MEDIA_TYPE_AUDIO))) { ast_log(LOG_NOTICE, "Asked to get a channel of unsupported format %s while capability is %s\n", - ast_getformatname_multiple(tmp, sizeof(tmp), cap), - ast_getformatname_multiple(tmp2, sizeof(tmp2), sip_cfg.caps)); + ast_format_cap_get_names(cap, &codec_buf), + ast_format_cap_get_names(sip_cfg.caps, &cap_buf)); *cause = AST_CAUSE_BEARERCAPABILITY_NOTAVAIL; /* Can't find codec to connect to host */ return NULL; } - ast_debug(1, "Asked to create a SIP channel with formats: %s\n", ast_getformatname_multiple(tmp, sizeof(tmp), cap)); + ast_debug(1, "Asked to create a SIP channel with formats: %s\n", ast_format_cap_get_names(cap, &codec_buf)); if (ast_strlen_zero(dest)) { ast_log(LOG_ERROR, "Unable to create channel with empty destination.\n"); @@ -29739,8 +29768,8 @@ static struct ast_channel *sip_request_call(const char *type, struct ast_format_ #if 0 printf("Setting up to call extension '%s' at '%s'\n", ext ? ext : "<none>", host); #endif - ast_format_cap_append(p->prefcaps, cap); - ast_format_cap_joint_copy(cap, p->caps, p->jointcaps); + ast_format_cap_append_from_cap(p->prefcaps, cap, AST_MEDIA_TYPE_UNKNOWN); + ast_format_cap_get_compatible(cap, p->caps, p->jointcaps); sip_pvt_lock(p); @@ -30217,7 +30246,7 @@ static void set_peer_defaults(struct sip_peer *peer) ast_string_field_set(peer, engine, default_engine); ast_sockaddr_setnull(&peer->addr); ast_sockaddr_setnull(&peer->defaddr); - ast_format_cap_copy(peer->caps, sip_cfg.caps); + ast_format_cap_append_from_cap(peer->caps, sip_cfg.caps, AST_MEDIA_TYPE_UNKNOWN); peer->maxcallbitrate = default_maxcallbitrate; peer->rtptimeout = global_rtptimeout; peer->rtpholdtimeout = global_rtpholdtimeout; @@ -30243,7 +30272,6 @@ static void set_peer_defaults(struct sip_peer *peer) peer->pickupgroup = 0; peer->maxms = default_qualify; peer->keepalive = default_keepalive; - peer->prefs = default_prefs; ast_string_field_set(peer, zone, default_zone); peer->stimer.st_mode_oper = global_st_mode; /* Session-Timers */ peer->stimer.st_ref = global_st_refresher; @@ -30279,7 +30307,7 @@ static struct sip_peer *temp_peer(const char *name) return NULL; } - if (!(peer->caps = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_NOLOCK))) { + if (!(peer->caps = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT))) { ao2_t_ref(peer, -1, "failed to allocate format capabilities, drop peer"); return NULL; } @@ -30291,7 +30319,6 @@ static struct sip_peer *temp_peer(const char *name) peer->selfdestruct = TRUE; peer->host_dynamic = TRUE; - peer->prefs = default_prefs; reg_source_db(peer); return peer; @@ -30384,7 +30411,7 @@ static struct sip_peer *build_peer(const char *name, struct ast_variable *v, str if (!(peer->endpoint = ast_endpoint_create("SIP", name))) { return NULL; } - if (!(peer->caps = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_NOLOCK))) { + if (!(peer->caps = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT))) { ao2_t_ref(peer, -1, "failed to allocate format capabilities, drop peer"); return NULL; } @@ -30705,12 +30732,12 @@ static struct sip_peer *build_peer(const char *name, struct ast_variable *v, str } else if (!strcasecmp(v->name, "namedpickupgroup")) { peer->named_pickupgroups = ast_get_namedgroups(v->value); } else if (!strcasecmp(v->name, "allow")) { - int error = ast_parse_allow_disallow(&peer->prefs, peer->caps, v->value, TRUE); + int error = ast_format_cap_update_by_allow_disallow(peer->caps, v->value, TRUE); if (error) { ast_log(LOG_WARNING, "Codec configuration errors found in line %d : %s = %s\n", v->lineno, v->name, v->value); } } else if (!strcasecmp(v->name, "disallow")) { - int error = ast_parse_allow_disallow(&peer->prefs, peer->caps, v->value, FALSE); + int error = ast_format_cap_update_by_allow_disallow(peer->caps, v->value, FALSE); if (error) { ast_log(LOG_WARNING, "Codec configuration errors found in line %d : %s = %s\n", v->lineno, v->name, v->value); } @@ -31104,14 +31131,11 @@ static int peer_markall_autopeers_func(void *device, void *arg, int flags) */ static void sip_set_default_format_capabilities(struct ast_format_cap *cap) { - struct ast_format tmp_fmt; - - ast_format_cap_remove_all(cap); - ast_format_cap_add(cap, ast_format_set(&tmp_fmt, AST_FORMAT_ULAW, 0)); - ast_format_cap_add(cap, ast_format_set(&tmp_fmt, AST_FORMAT_TESTLAW, 0)); - ast_format_cap_add(cap, ast_format_set(&tmp_fmt, AST_FORMAT_ALAW, 0)); - ast_format_cap_add(cap, ast_format_set(&tmp_fmt, AST_FORMAT_GSM, 0)); - ast_format_cap_add(cap, ast_format_set(&tmp_fmt, AST_FORMAT_H263, 0)); + ast_format_cap_remove_by_type(cap, AST_MEDIA_TYPE_UNKNOWN); + ast_format_cap_append(cap, ast_format_ulaw, 0); + ast_format_cap_append(cap, ast_format_alaw, 0); + ast_format_cap_append(cap, ast_format_gsm, 0); + ast_format_cap_append(cap, ast_format_h263, 0); } static void display_nat_warning(const char *cat, int reason, struct ast_flags *flags) { @@ -31276,7 +31300,6 @@ static int reload_config(enum channelreloadreason reason) memset(&localaddr, 0, sizeof(localaddr)); memset(&externaddr, 0, sizeof(externaddr)); memset(&media_address, 0, sizeof(media_address)); - memset(&default_prefs, 0 , sizeof(default_prefs)); memset(&sip_cfg.outboundproxy, 0, sizeof(struct sip_proxy)); sip_cfg.outboundproxy.force = FALSE; /*!< Don't force proxy usage, use route: headers */ default_transports = AST_TRANSPORT_UDP; @@ -31768,12 +31791,12 @@ static int reload_config(enum channelreloadreason reason) ast_log(LOG_WARNING, "Invalid externtlsport value, must be a positive integer between 1 and 65535 at line %d\n", v->lineno); } } else if (!strcasecmp(v->name, "allow")) { - int error = ast_parse_allow_disallow(&default_prefs, sip_cfg.caps, v->value, TRUE); + int error = ast_format_cap_update_by_allow_disallow(sip_cfg.caps, v->value, TRUE); if (error) { ast_log(LOG_WARNING, "Codec configuration errors found in line %d : %s = %s\n", v->lineno, v->name, v->value); } } else if (!strcasecmp(v->name, "disallow")) { - int error = ast_parse_allow_disallow(&default_prefs, sip_cfg.caps, v->value, FALSE); + int error = ast_format_cap_update_by_allow_disallow(sip_cfg.caps, v->value, FALSE); if (error) { ast_log(LOG_WARNING, "Codec configuration errors found in line %d : %s = %s\n", v->lineno, v->name, v->value); } @@ -32565,8 +32588,9 @@ static int sip_set_rtp_peer(struct ast_channel *chan, struct ast_rtp_instance *i memset(&p->tredirip, 0, sizeof(p->tredirip)); changed = 1; } - if (cap && !ast_format_cap_is_empty(cap) && !ast_format_cap_identical(p->redircaps, cap)) { - ast_format_cap_copy(p->redircaps, cap); + if (cap && ast_format_cap_count(cap) && !ast_format_cap_identical(cap, p->redircaps)) { + ast_format_cap_remove_by_type(p->redircaps, AST_MEDIA_TYPE_UNKNOWN); + ast_format_cap_append_from_cap(p->redircaps, cap, AST_MEDIA_TYPE_UNKNOWN); changed = 1; } @@ -32603,7 +32627,8 @@ static int sip_set_rtp_peer(struct ast_channel *chan, struct ast_rtp_instance *i static void sip_get_codec(struct ast_channel *chan, struct ast_format_cap *result) { struct sip_pvt *p = ast_channel_tech_pvt(chan); - ast_format_cap_append(result, ast_format_cap_is_empty(p->peercaps) ? p->caps : p->peercaps); + + ast_format_cap_append_from_cap(result, !ast_format_cap_count(p->peercaps) ? p->caps : p->peercaps, AST_MEDIA_TYPE_UNKNOWN); } static struct ast_rtp_glue sip_rtp_glue = { @@ -34217,6 +34242,8 @@ static const struct ast_sip_api_tech chan_sip_api_provider = { .sipinfo_send = sipinfo_send, }; +static int unload_module(void); + /*! * \brief Load the module * @@ -34232,14 +34259,17 @@ static int load_module(void) ast_verbose("SIP channel loading...\n"); if (STASIS_MESSAGE_TYPE_INIT(session_timeout_type)) { + unload_module(); return AST_MODULE_LOAD_FAILURE; } if (!(sip_tech.capabilities = ast_format_cap_alloc(0))) { + unload_module(); return AST_MODULE_LOAD_FAILURE; } if (ast_sip_api_provider_register(&chan_sip_api_provider)) { + unload_module(); return AST_MODULE_LOAD_FAILURE; } @@ -34254,24 +34284,28 @@ static int load_module(void) if (!peers || !peers_by_ip || !dialogs || !dialogs_needdestroy || !dialogs_rtpcheck || !threadt) { ast_log(LOG_ERROR, "Unable to create primary SIP container(s)\n"); + unload_module(); return AST_MODULE_LOAD_FAILURE; } - if (!(sip_cfg.caps = ast_format_cap_alloc(0))) { + if (!(sip_cfg.caps = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT))) { + unload_module(); return AST_MODULE_LOAD_FAILURE; } - ast_format_cap_add_all_by_type(sip_tech.capabilities, AST_FORMAT_TYPE_AUDIO); + ast_format_cap_append_by_type(sip_tech.capabilities, AST_MEDIA_TYPE_AUDIO); ASTOBJ_CONTAINER_INIT(®l); /* Registry object list -- not searched for anything */ ASTOBJ_CONTAINER_INIT(&submwil); /* MWI subscription object list */ if (!(sched = ast_sched_context_create())) { ast_log(LOG_ERROR, "Unable to create scheduler context\n"); + unload_module(); return AST_MODULE_LOAD_FAILURE; } if (!(io = io_context_create())) { ast_log(LOG_ERROR, "Unable to create I/O context\n"); + unload_module(); return AST_MODULE_LOAD_FAILURE; } @@ -34279,15 +34313,14 @@ static int load_module(void) can_parse_xml = sip_is_xml_parsable(); if (reload_config(sip_reloadreason)) { /* Load the configuration from sip.conf */ - ast_sip_api_provider_unregister(); + unload_module(); return AST_MODULE_LOAD_DECLINE; } /* Initialize bogus peer. Can be done first after reload_config() */ if (!(bogus_peer = temp_peer("(bogus_peer)"))) { ast_log(LOG_ERROR, "Unable to create bogus_peer for authentication\n"); - io_context_destroy(io); - ast_sched_context_destroy(sched); + unload_module(); return AST_MODULE_LOAD_FAILURE; } /* Make sure the auth will always fail. */ @@ -34302,16 +34335,14 @@ static int load_module(void) memset((void *) &sip_tech_info.send_digit_begin, 0, sizeof(sip_tech_info.send_digit_begin)); if (ast_msg_tech_register(&sip_msg_tech)) { - /* LOAD_FAILURE stops Asterisk, so cleanup is a moot point. */ + unload_module(); return AST_MODULE_LOAD_FAILURE; } /* Make sure we can register our sip channel type */ if (ast_channel_register(&sip_tech)) { ast_log(LOG_ERROR, "Unable to register channel type 'SIP'\n"); - ao2_t_ref(bogus_peer, -1, "unref the bogus_peer"); - io_context_destroy(io); - ast_sched_context_destroy(sched); + unload_module(); return AST_MODULE_LOAD_FAILURE; } @@ -34358,13 +34389,13 @@ static int load_module(void) initialize_escs(); if (sip_epa_register(&cc_epa_static_data)) { - ast_sip_api_provider_unregister(); + unload_module(); return AST_MODULE_LOAD_DECLINE; } if (sip_reqresp_parser_init() == -1) { ast_log(LOG_ERROR, "Unable to initialize the SIP request and response parser\n"); - ast_sip_api_provider_unregister(); + unload_module(); return AST_MODULE_LOAD_DECLINE; } @@ -34373,16 +34404,16 @@ static int load_module(void) * in incoming PUBLISH requests */ if (ast_cc_agent_register(&sip_cc_agent_callbacks)) { - ast_sip_api_provider_unregister(); + unload_module(); return AST_MODULE_LOAD_DECLINE; } } if (ast_cc_monitor_register(&sip_cc_monitor_callbacks)) { - ast_sip_api_provider_unregister(); + unload_module(); return AST_MODULE_LOAD_DECLINE; } if (!(sip_monitor_instances = ao2_container_alloc(37, sip_monitor_instance_hash_fn, sip_monitor_instance_cmp_fn))) { - ast_sip_api_provider_unregister(); + unload_module(); return AST_MODULE_LOAD_DECLINE; } @@ -34523,7 +34554,7 @@ static int unload_module(void) ast_mutex_lock(&authl_lock); if (authl) { - ao2_t_ref(authl, -1, "Removing global authentication"); + ao2_t_cleanup(authl, "Removing global authentication"); authl = NULL; } ast_mutex_unlock(&authl_lock); @@ -34568,15 +34599,15 @@ static int unload_module(void) ast_debug(2, "TCP/TLS thread container did not become empty :(\n"); } - ao2_t_ref(bogus_peer, -1, "unref the bogus_peer"); + ao2_t_cleanup(bogus_peer, "unref the bogus_peer"); - ao2_t_ref(peers, -1, "unref the peers table"); - ao2_t_ref(peers_by_ip, -1, "unref the peers_by_ip table"); - ao2_t_ref(dialogs, -1, "unref the dialogs table"); - ao2_t_ref(dialogs_needdestroy, -1, "unref dialogs_needdestroy"); - ao2_t_ref(dialogs_rtpcheck, -1, "unref dialogs_rtpcheck"); - ao2_t_ref(threadt, -1, "unref the thread table"); - ao2_t_ref(sip_monitor_instances, -1, "unref the sip_monitor_instances table"); + ao2_t_cleanup(peers, "unref the peers table"); + ao2_t_cleanup(peers_by_ip, "unref the peers_by_ip table"); + ao2_t_cleanup(dialogs, "unref the dialogs table"); + ao2_t_cleanup(dialogs_needdestroy, "unref dialogs_needdestroy"); + ao2_t_cleanup(dialogs_rtpcheck, "unref dialogs_rtpcheck"); + ao2_t_cleanup(threadt, "unref the thread table"); + ao2_t_cleanup(sip_monitor_instances, "unref the sip_monitor_instances table"); clear_sip_domains(); sip_cfg.contact_acl = ast_free_acl_list(sip_cfg.contact_acl); @@ -34604,8 +34635,10 @@ static int unload_module(void) notify_types = NULL; } - ast_format_cap_destroy(sip_tech.capabilities); - sip_cfg.caps = ast_format_cap_destroy(sip_cfg.caps); + ao2_cleanup(sip_tech.capabilities); + sip_tech.capabilities = NULL; + ao2_cleanup(sip_cfg.caps); + sip_cfg.caps = NULL; STASIS_MESSAGE_TYPE_CLEANUP(session_timeout_type); |