diff options
Diffstat (limited to 'channels/chan_sip.c')
| -rw-r--r-- | channels/chan_sip.c | 627 |
1 files changed, 379 insertions, 248 deletions
diff --git a/channels/chan_sip.c b/channels/chan_sip.c index d5ee01345..e80f387cc 100644 --- a/channels/chan_sip.c +++ b/channels/chan_sip.c @@ -1199,7 +1199,7 @@ static struct ast_config *notify_types = NULL; /*!< The list of manual NOTIFY in coming releases. */ /*--- PBX interface functions */ -static struct ast_channel *sip_request_call(const char *type, format_t format, const struct ast_channel *requestor, void *data, int *cause); +static struct ast_channel *sip_request_call(const char *type, struct ast_format_cap *cap, const struct ast_channel *requestor, void *data, int *cause); static int sip_devicestate(void *data); static int sip_sendtext(struct ast_channel *ast, const char *text); static int sip_call(struct ast_channel *ast, char *dest, int timeout); @@ -1291,7 +1291,7 @@ static int process_sdp_a_audio(const char *a, struct sip_pvt *p, struct ast_rtp_ static int process_sdp_a_video(const char *a, struct sip_pvt *p, struct ast_rtp_codecs *newvideortp, int *last_rtpmap_codec); static int process_sdp_a_text(const char *a, struct sip_pvt *p, struct ast_rtp_codecs *newtextrtp, char *red_fmtp, int *red_num_gen, int *red_data_pt, int *last_rtpmap_codec); static int process_sdp_a_image(const char *a, struct sip_pvt *p); -static void add_codec_to_sdp(const struct sip_pvt *p, format_t codec, +static void add_codec_to_sdp(const struct sip_pvt *p, struct ast_format *codec, struct ast_str **m_buf, struct ast_str **a_buf, int debug, int *min_packet_size); static void add_noncodec_to_sdp(const struct sip_pvt *p, int format, @@ -1565,7 +1565,7 @@ static enum st_mode st_get_mode(struct sip_pvt *); static struct sip_st_dlg* sip_st_alloc(struct sip_pvt *const p); /*------- RTP Glue functions -------- */ -static int sip_set_rtp_peer(struct ast_channel *chan, struct ast_rtp_instance *instance, struct ast_rtp_instance *vinstance, struct ast_rtp_instance *tinstance, format_t codecs, int nat_active); +static int sip_set_rtp_peer(struct ast_channel *chan, struct ast_rtp_instance *instance, struct ast_rtp_instance *vinstance, struct ast_rtp_instance *tinstance, const struct ast_format_cap *cap, int nat_active); /*!--- SIP MWI Subscription support */ static int sip_subscribe_mwi(const char *value, int lineno); @@ -1575,10 +1575,9 @@ static int sip_subscribe_mwi_do(const void *data); static int __sip_subscribe_mwi_do(struct sip_subscription_mwi *mwi); /*! \brief Definition of this channel for PBX channel registration */ -const struct ast_channel_tech sip_tech = { +struct ast_channel_tech sip_tech = { .type = "SIP", .description = "Session Initiation Protocol (SIP)", - .capabilities = AST_FORMAT_AUDIO_MASK, /* all audio formats */ .properties = AST_CHAN_TP_WANTSJITTER | AST_CHAN_TP_CREATESJITTER, .requester = sip_request_call, /* called with chan unlocked */ .devicestate = sip_devicestate, /* called with chan unlocked (not chan-specific) */ @@ -4038,10 +4037,10 @@ static int sip_setoption(struct ast_channel *chan, int option, void *data, int d switch (option) { case AST_OPTION_FORMAT_READ: - res = ast_rtp_instance_set_read_format(p->rtp, *(int *) data); + res = ast_rtp_instance_set_read_format(p->rtp, (struct ast_format *) data); break; case AST_OPTION_FORMAT_WRITE: - res = ast_rtp_instance_set_write_format(p->rtp, *(int *) data); + res = ast_rtp_instance_set_write_format(p->rtp, (struct ast_format *) data); break; case AST_OPTION_MAKE_COMPATIBLE: res = ast_rtp_instance_make_compatible(chan, p->rtp, (struct ast_channel *) data); @@ -4385,6 +4384,8 @@ static void sip_destroy_peer(struct sip_peer *peer) ast_cc_config_params_destroy(peer->cc_params); ast_string_field_free_memory(peer); + + peer->caps = ast_format_cap_destroy(peer->caps); } /*! \brief Update peer data in database (if used) */ @@ -4826,7 +4827,7 @@ 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) && (dialog->capability & AST_FORMAT_VIDEO_MASK))) { + (ast_test_flag(&dialog->flags[1], SIP_PAGE2_VIDEOSUPPORT) && (ast_format_cap_has_type(dialog->caps, AST_FORMAT_TYPE_VIDEO)))) { if (!(dialog->vrtp = ast_rtp_instance_new(dialog->engine, sched, &bindaddr_tmp, NULL))) { return -1; } @@ -4887,7 +4888,7 @@ 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); - dialog->capability = peer->capability; + ast_format_cap_append(dialog->caps, peer->caps); dialog->prefs = peer->prefs; if (ast_test_flag(&dialog->flags[1], SIP_PAGE2_T38SUPPORT)) { /* t38pt_udptl was enabled in the peer and not in [general] */ @@ -5253,11 +5254,11 @@ static int sip_call(struct ast_channel *ast, char *dest, int timeout) return res; } p->callingpres = ast_party_id_presentation(&ast->caller.id); - p->jointcapability = ast_rtp_instance_available_formats(p->rtp, p->capability, p->prefcodec); + ast_rtp_instance_available_formats(p->rtp, p->caps, p->prefcaps, p->jointcaps); p->jointnoncodeccapability = p->noncodeccapability; /* If there are no audio formats left to offer, punt */ - if (!(p->jointcapability & AST_FORMAT_AUDIO_MASK)) { + if (!(ast_format_cap_has_type(p->jointcaps, AST_FORMAT_TYPE_AUDIO))) { ast_log(LOG_WARNING, "No audio format found to offer. Cancelling call to %s\n", p->username); res = -1; } else { @@ -5463,6 +5464,11 @@ void __sip_destroy(struct sip_pvt *p, int lockowner, int lockdialoglist) ao2_ref(p->socket.tcptls_session, -1); p->socket.tcptls_session = NULL; } + 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); } /*! \brief update_call_counter: Handle call_limit for SIP devices @@ -6025,9 +6031,11 @@ static int sip_hangup(struct ast_channel *ast) /*! \brief Try setting codec suggested by the SIP_CODEC channel variable */ static void try_suggested_sip_codec(struct sip_pvt *p) { - format_t fmt; + struct ast_format fmt; const char *codec; + ast_format_clear(&fmt); + if (p->outgoing_call) { codec = pbx_builtin_getvar_helper(p->owner, "SIP_CODEC_OUTBOUND"); } else if (!(codec = pbx_builtin_getvar_helper(p->owner, "SIP_CODEC_INBOUND"))) { @@ -6037,12 +6045,12 @@ static void try_suggested_sip_codec(struct sip_pvt *p) if (!codec) return; - fmt = ast_getformatbyname(codec); - if (fmt) { + ast_getformatbyname(codec, &fmt); + if (fmt.id) { ast_log(LOG_NOTICE, "Changing codec to '%s' for this call because of ${SIP_CODEC} variable\n", codec); - if (p->jointcapability & fmt) { - p->jointcapability &= fmt; - p->capability &= fmt; + if (ast_format_cap_iscompatible(p->jointcaps, &fmt)) { + ast_format_cap_set(p->jointcaps, &fmt); + ast_format_cap_set(p->caps, &fmt); } else ast_log(LOG_NOTICE, "Ignoring ${SIP_CODEC} variable because it is not shared by both ends.\n"); } else @@ -6079,13 +6087,13 @@ static int sip_write(struct ast_channel *ast, struct ast_frame *frame) switch (frame->frametype) { case AST_FRAME_VOICE: - if (!(frame->subclass.codec & ast->nativeformats)) { - char s1[512], s2[512], s3[512]; + if (!(ast_format_cap_iscompatible(ast->nativeformats, &frame->subclass.format))) { + char s1[512]; ast_log(LOG_WARNING, "Asked to transmit frame type %s, while native formats is %s read/write = %s/%s\n", - ast_getformatname(frame->subclass.codec), - ast_getformatname_multiple(s1, sizeof(s1), ast->nativeformats & AST_FORMAT_AUDIO_MASK), - ast_getformatname_multiple(s2, sizeof(s2), ast->readformat), - ast_getformatname_multiple(s3, sizeof(s3), ast->writeformat)); + ast_getformatname(&frame->subclass.format), + ast_getformatname_multiple(s1, sizeof(s1), ast->nativeformats), + ast_getformatname(&ast->readformat), + ast_getformatname(&ast->writeformat)); return 0; } if (p) { @@ -6538,11 +6546,9 @@ static struct ast_channel *sip_new(struct sip_pvt *i, int state, const char *tit { struct ast_channel *tmp; struct ast_variable *v = NULL; - format_t fmt; - format_t what; - format_t video; - format_t text; - format_t needvideo = 0; + struct ast_format fmt; + struct ast_format_cap *what = NULL; /* SHALLOW COPY DO NOT DESTROY! */ + int needvideo = 0; int needtext = 0; char buf[SIPBUFSIZE]; char *decoded_exten; @@ -6575,51 +6581,47 @@ 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 (i->jointcapability) { /* The joint capabilities of us and peer */ - what = i->jointcapability; - video = i->jointcapability & AST_FORMAT_VIDEO_MASK; - text = i->jointcapability & AST_FORMAT_TEXT_MASK; - } else if (i->capability) { /* Our configured capability for this peer */ - what = i->capability; - video = i->capability & AST_FORMAT_VIDEO_MASK; - text = i->capability & AST_FORMAT_TEXT_MASK; + if (!(ast_format_cap_is_empty(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 */ + what = i->caps; } else { - what = sip_cfg.capability; /* Global codec support */ - video = sip_cfg.capability & AST_FORMAT_VIDEO_MASK; - text = sip_cfg.capability & AST_FORMAT_TEXT_MASK; + what = sip_cfg.caps; } - /* Set the native formats for audio and merge in video */ - tmp->nativeformats = ast_codec_choose(&i->prefs, what, 1) | video | text; + /* Set the native formats */ + ast_format_cap_copy(tmp->nativeformats, what); + /* choose and use only the best audio format for our native formats */ + ast_codec_choose(&i->prefs, tmp->nativeformats, 1, &fmt); /* get the best audio format */ + ast_format_cap_remove_bytype(tmp->nativeformats, AST_FORMAT_TYPE_AUDIO); /* remove only the other audio formats */ + ast_format_cap_add(tmp->nativeformats, &fmt); /* add our best choice back */ + ast_debug(3, "*** Our native formats are %s \n", ast_getformatname_multiple(buf, SIPBUFSIZE, tmp->nativeformats)); - ast_debug(3, "*** Joint capabilities are %s \n", ast_getformatname_multiple(buf, SIPBUFSIZE, i->jointcapability)); - ast_debug(3, "*** Our capabilities are %s \n", ast_getformatname_multiple(buf, SIPBUFSIZE, i->capability)); - ast_debug(3, "*** AST_CODEC_CHOOSE formats are %s \n", ast_getformatname_multiple(buf, SIPBUFSIZE, ast_codec_choose(&i->prefs, what, 1))); - if (i->prefcodec) { - ast_debug(3, "*** Our preferred formats from the incoming channel are %s \n", ast_getformatname_multiple(buf, SIPBUFSIZE, i->prefcodec)); + 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)); } - /* XXX Why are we choosing a codec from the native formats?? */ - 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 (ast_test_flag(&i->flags[1], SIP_PAGE2_VIDEOSUPPORT)) - needvideo = AST_FORMAT_VIDEO_MASK; - else if (i->prefcodec) - needvideo = i->prefcodec & AST_FORMAT_VIDEO_MASK; /* Outbound call */ + 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 - needvideo = i->jointcapability & AST_FORMAT_VIDEO_MASK; /* Inbound call */ + needvideo = ast_format_cap_has_type(i->jointcaps, AST_FORMAT_TYPE_VIDEO); /* Inbound call */ } if (i->trtp) { - if (i->prefcodec) - needtext = i->prefcodec & AST_FORMAT_TEXT_MASK; /* Outbound call */ + if (!ast_format_cap_is_empty(i->prefcaps)) + needtext = ast_format_cap_has_type(i->prefcaps, AST_FORMAT_TYPE_TEXT); /* Outbound call */ else - needtext = i->jointcapability & AST_FORMAT_TEXT_MASK; /* Inbound call */ + needtext = ast_format_cap_has_type(i->jointcaps, AST_FORMAT_TYPE_TEXT); /* Inbound call */ } if (needvideo) { @@ -6662,13 +6664,13 @@ static struct ast_channel *sip_new(struct sip_pvt *i, int state, const char *tit } tmp->adsicpe = AST_ADSI_UNAVAILABLE; - tmp->writeformat = fmt; - tmp->rawwriteformat = fmt; - ast_rtp_instance_set_write_format(i->rtp, fmt); + ast_format_copy(&tmp->writeformat, &fmt); + ast_format_copy(&tmp->rawwriteformat, &fmt); + ast_rtp_instance_set_write_format(i->rtp, &fmt); - tmp->readformat = fmt; - tmp->rawreadformat = fmt; - ast_rtp_instance_set_read_format(i->rtp, fmt); + ast_format_copy(&tmp->readformat, &fmt); + ast_format_copy(&tmp->rawreadformat, &fmt); + ast_rtp_instance_set_read_format(i->rtp, &fmt); tmp->tech_pvt = dialog_ref(i, "sip_new: set chan->tech_pvt to i"); @@ -6976,17 +6978,18 @@ static struct ast_frame *sip_rtp_read(struct ast_channel *ast, struct sip_pvt *p return f; } - if (f && f->subclass.codec != (p->owner->nativeformats & AST_FORMAT_AUDIO_MASK)) { - if (!(f->subclass.codec & p->jointcapability)) { + if (f && !ast_format_cap_iscompatible(p->owner->nativeformats, &f->subclass.format)) { + if (!ast_format_cap_iscompatible(p->jointcaps, &f->subclass.format)) { ast_debug(1, "Bogus frame of format '%s' received from '%s'!\n", - ast_getformatname(f->subclass.codec), p->owner->name); + ast_getformatname(&f->subclass.format), p->owner->name); return &ast_null_frame; } ast_debug(1, "Oooh, format changed to %s\n", - ast_getformatname(f->subclass.codec)); - p->owner->nativeformats = (p->owner->nativeformats & (AST_FORMAT_VIDEO_MASK | AST_FORMAT_TEXT_MASK)) | f->subclass.codec; - ast_set_read_format(p->owner, p->owner->readformat); - ast_set_write_format(p->owner, p->owner->writeformat); + ast_getformatname(&f->subclass.format)); + ast_format_cap_remove_bytype(p->owner->nativeformats, AST_FORMAT_TYPE_AUDIO); + ast_format_cap_add(p->owner->nativeformats, &f->subclass.format); + ast_set_read_format(p->owner, &p->owner->readformat); + ast_set_write_format(p->owner, &p->owner->writeformat); } if (f && p->dsp) { @@ -7149,6 +7152,22 @@ struct sip_pvt *sip_alloc(ast_string_field callid, struct ast_sockaddr *addr, ao2_t_ref(p, -1, "Yuck, couldn't allocate cc_params struct. Get rid o' p"); return NULL; } + p->caps = ast_format_cap_alloc_nolock(); + p->jointcaps = ast_format_cap_alloc_nolock(); + p->peercaps = ast_format_cap_alloc_nolock(); + p->redircaps = ast_format_cap_alloc_nolock(); + p->prefcaps = ast_format_cap_alloc_nolock(); + + 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); + ao2_t_ref(p, -1, "Yuck, couldn't allocate cc_params struct. Get rid o' p"); + return NULL; + } + /* If this dialog is created as a result of a request or response, lets store * some information about it in the dialog. */ @@ -7255,7 +7274,7 @@ 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); - p->capability = sip_cfg.capability; + ast_format_cap_append(p->caps, sip_cfg.caps); 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)) @@ -8262,6 +8281,8 @@ static int get_ip_and_port_from_sdp(struct sip_request *req, const enum media_ty */ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action) { + int res = 0; + /* Iterators for SDP parsing */ int start = req->sdp_start; int next = start; @@ -8284,18 +8305,21 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action struct ast_sockaddr *vsa = NULL; /*!< RTP video host IP */ struct ast_sockaddr *tsa = NULL; /*!< RTP text host IP */ struct ast_sockaddr *isa = NULL; /*!< UDPTL host ip */ - int portno = -1; /*!< RTP Audio port number */ - int vportno = -1; /*!< RTP Video port number */ + int portno = -1; /*!< RTP Audio port number */ + int vportno = -1; /*!< RTP Video port number */ int tportno = -1; /*!< RTP Text port number */ int udptlportno = -1; /*!< UDPTL Image port number */ /* Peer capability is the capability in the SDP, non codec is RFC2833 DTMF (101) */ - format_t peercapability = 0, vpeercapability = 0, tpeercapability = 0; + struct ast_format_cap *peercapability = ast_format_cap_alloc_nolock(); + struct ast_format_cap *vpeercapability = ast_format_cap_alloc_nolock(); + struct ast_format_cap *tpeercapability = ast_format_cap_alloc_nolock(); + int peernoncodeccapability = 0, vpeernoncodeccapability = 0, tpeernoncodeccapability = 0; struct ast_rtp_codecs newaudiortp, newvideortp, newtextrtp; - format_t newjointcapability; /* Negotiated capability */ - format_t newpeercapability; + struct ast_format_cap *newjointcapability = ast_format_cap_alloc_nolock(); /* Negotiated capability */ + struct ast_format_cap *newpeercapability = ast_format_cap_alloc_nolock(); int newnoncodeccapability; const char *codecs; @@ -8318,12 +8342,18 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action /* START UNKNOWN */ char buf[SIPBUFSIZE]; + struct ast_format tmp_fmt; /* END UNKNOWN */ /* Initial check */ if (!p->rtp) { ast_log(LOG_ERROR, "Got SDP but have no RTP session allocated.\n"); - return -1; + res = -1; + goto process_sdp_cleanup; + } + if (!peercapability || !vpeercapability || !tpeercapability || !newpeercapability || !newjointcapability) { + res = -1; + goto process_sdp_cleanup; } /* Make sure that the codec structures are all cleared out */ @@ -8353,7 +8383,8 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action nextm = get_sdp_iterate(&next, req, "m"); if (ast_strlen_zero(nextm)) { ast_log(LOG_WARNING, "Insufficient information for SDP (m= not found)\n"); - return -1; + res = -1; + goto process_sdp_cleanup; } /* Scan session level SDP parameters (lines before first media stream) */ @@ -8364,8 +8395,10 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action /* If we end up receiving SDP that doesn't actually modify the session we don't want to treat this as a fatal * error. We just want to ignore the SDP and let the rest of the packet be handled as normal. */ - if (!process_sdp_o(value, p)) - return (p->session_modify == FALSE) ? 0 : -1; + if (!process_sdp_o(value, p)) { + res = (p->session_modify == FALSE) ? 0 : -1; + goto process_sdp_cleanup; + } break; case 'c': if (process_sdp_c(value, &sessionsa)) { @@ -8433,7 +8466,8 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action for (; !ast_strlen_zero(codecs); codecs = ast_skip_blanks(codecs + len)) { if (sscanf(codecs, "%30u%n", &codec, &len) != 1) { ast_log(LOG_WARNING, "Error in codec string '%s'\n", codecs); - return -1; + res = -1; + goto process_sdp_cleanup; } if (debug) ast_verbose("Found RTP audio format %d\n", codec); @@ -8461,7 +8495,8 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action for (; !ast_strlen_zero(codecs); codecs = ast_skip_blanks(codecs + len)) { if (sscanf(codecs, "%30u%n", &codec, &len) != 1) { ast_log(LOG_WARNING, "Error in codec string '%s'\n", codecs); - return -1; + res = -1; + goto process_sdp_cleanup; } if (debug) ast_verbose("Found RTP video format %d\n", codec); @@ -8482,7 +8517,8 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action for (; !ast_strlen_zero(codecs); codecs = ast_skip_blanks(codecs + len)) { if (sscanf(codecs, "%30u%n", &codec, &len) != 1) { ast_log(LOG_WARNING, "Error in codec string '%s'\n", codecs); - return -1; + res = -1; + goto process_sdp_cleanup; } if (debug) ast_verbose("Found RTP text format %d\n", codec); @@ -8587,45 +8623,53 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action /* Sanity checks */ if (!sa && !vsa && !tsa && !isa) { ast_log(LOG_WARNING, "Insufficient information in SDP (c=)...\n"); - return -1; + res = -1; + goto process_sdp_cleanup; } if (portno == -1 && vportno == -1 && udptlportno == -1 && tportno == -1) { /* No acceptable offer found in SDP - we have no ports */ /* Do not change RTP or VRTP if this is a re-invite */ ast_log(LOG_WARNING, "Failing due to no acceptable offer found\n"); - return -2; + res = -2; + goto process_sdp_cleanup; } if (numberofmediastreams > 3) { /* We have too many fax, audio and/or video and/or text media streams, fail this offer */ ast_log(LOG_WARNING, "Faling due to too many media streams\n"); - return -3; + res = -3; + goto process_sdp_cleanup; } if (secure_audio && !(p->srtp && (ast_test_flag(p->srtp, SRTP_CRYPTO_OFFER_OK)))) { ast_log(LOG_WARNING, "Can't provide secure audio requested in SDP offer\n"); - return -4; + res = -4; + goto process_sdp_cleanup; } if (!secure_audio && p->srtp) { ast_log(LOG_WARNING, "We are requesting SRTP, but they responded without it!\n"); - return -4; + res = -4; + goto process_sdp_cleanup; } if (secure_video && !(p->vsrtp && (ast_test_flag(p->vsrtp, SRTP_CRYPTO_OFFER_OK)))) { ast_log(LOG_WARNING, "Can't provide secure video requested in SDP offer\n"); - return -4; + res = -4; + goto process_sdp_cleanup; } if (!p->novideo && !secure_video && p->vsrtp) { ast_log(LOG_WARNING, "We are requesting SRTP, but they responded without it!\n"); - return -4; + res = -4; + goto process_sdp_cleanup; } if (!(secure_audio || secure_video) && ast_test_flag(&p->flags[1], SIP_PAGE2_USE_SRTP)) { ast_log(LOG_WARNING, "Matched device setup to use SRTP, but request was not!\n"); - return -4; + res = -4; + goto process_sdp_cleanup; } if (udptlportno == -1) { @@ -8633,12 +8677,22 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action } /* Now gather all of the codecs that we are asked for: */ - ast_rtp_codecs_payload_formats(&newaudiortp, &peercapability, &peernoncodeccapability); - ast_rtp_codecs_payload_formats(&newvideortp, &vpeercapability, &vpeernoncodeccapability); - ast_rtp_codecs_payload_formats(&newtextrtp, &tpeercapability, &tpeernoncodeccapability); + ast_rtp_codecs_payload_formats(&newaudiortp, peercapability, &peernoncodeccapability); + 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_joint_copy(p->caps, newpeercapability, newjointcapability); + if (ast_format_cap_is_empty(newjointcapability) && (portno != -1)) { + ast_log(LOG_NOTICE, "No compatible codecs, not accepting this offer!\n"); + /* Do NOT Change current setting */ + res = -1; + goto process_sdp_cleanup; + } - newjointcapability = p->capability & (peercapability | vpeercapability | tpeercapability); - newpeercapability = (peercapability | vpeercapability | tpeercapability); newnoncodeccapability = p->noncodeccapability & peernoncodeccapability; if (debug) { @@ -8646,7 +8700,7 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action char s1[SIPBUFSIZE], s2[SIPBUFSIZE], s3[SIPBUFSIZE], s4[SIPBUFSIZE], s5[SIPBUFSIZE]; ast_verbose("Capabilities: us - %s, peer - audio=%s/video=%s/text=%s, combined - %s\n", - ast_getformatname_multiple(s1, SIPBUFSIZE, p->capability), + 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), @@ -8658,14 +8712,9 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action struct ast_str *s3 = ast_str_alloca(SIPBUFSIZE); ast_verbose("Non-codec capabilities (dtmf): us - %s, peer - %s, combined - %s\n", - ast_rtp_lookup_mime_multiple2(s1, p->noncodeccapability, 0, 0), - ast_rtp_lookup_mime_multiple2(s2, peernoncodeccapability, 0, 0), - ast_rtp_lookup_mime_multiple2(s3, newnoncodeccapability, 0, 0)); - } - if (!newjointcapability && (portno != -1)) { - ast_log(LOG_NOTICE, "No compatible codecs, not accepting this offer!\n"); - /* Do NOT Change current setting */ - return -1; + ast_rtp_lookup_mime_multiple2(s1, NULL, p->noncodeccapability, 0, 0), + ast_rtp_lookup_mime_multiple2(s2, NULL, peernoncodeccapability, 0, 0), + ast_rtp_lookup_mime_multiple2(s3, NULL, newnoncodeccapability, 0, 0)); } /* Setup audio address and port */ @@ -8679,12 +8728,13 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action } /* 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 */ + ast_format_cap_copy(p->jointcaps, newjointcapability); /* Our joint codec profile for this call */ + ast_format_cap_copy(p->peercaps, newpeercapability); /* The other sides capability in latest offer */ p->jointnoncodeccapability = newnoncodeccapability; /* DTMF capabilities */ if (ast_test_flag(&p->flags[1], SIP_PAGE2_PREFERRED_CODEC)) { /* respond with single most preferred joint codec, limiting the other side's choice */ - p->jointcapability = ast_codec_choose(&p->prefs, p->jointcapability, 1); + ast_codec_choose(&p->prefs, p->jointcaps, 1, &tmp_fmt); + ast_format_cap_set(p->jointcaps, &tmp_fmt); } ast_rtp_codecs_payloads_copy(&newaudiortp, ast_rtp_instance_get_codecs(p->rtp), p->rtp); @@ -8737,7 +8787,7 @@ 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 ((p->jointcapability & AST_FORMAT_T140RED)) { + if (ast_format_cap_iscompatible(p->jointcaps, ast_format_set(&tmp_fmt, AST_FORMAT_T140RED, 0))) { p->red = 1; ast_rtp_red_init(p->trtp, 300, red_data_pt, 2); } else { @@ -8807,27 +8857,36 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action if ((portno == -1) && (p->t38.state != T38_DISABLED)) { ast_debug(3, "Have T.38 but no audio, accepting offer anyway\n"); - return 0; - } + res = 0; + goto process_sdp_cleanup; + } /* Ok, we're going with this offer */ - ast_debug(2, "We're settling with these formats: %s\n", ast_getformatname_multiple(buf, SIPBUFSIZE, p->jointcapability)); + ast_debug(2, "We're settling with these formats: %s\n", ast_getformatname_multiple(buf, SIPBUFSIZE, p->jointcaps)); - 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; + 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; + goto process_sdp_cleanup; + } ast_debug(4, "We have an owner, now see if we need to change this call\n"); - if (!(p->owner->nativeformats & p->jointcapability) && (p->jointcapability & AST_FORMAT_AUDIO_MASK)) { + if (!(ast_format_cap_has_joint(p->owner->nativeformats, p->jointcaps)) && ast_format_cap_has_type(p->jointcaps, AST_FORMAT_TYPE_AUDIO)) { if (debug) { char s1[SIPBUFSIZE], s2[SIPBUFSIZE]; ast_debug(1, "Oooh, we need to change our audio formats since our peer supports only %s and not %s\n", - ast_getformatname_multiple(s1, SIPBUFSIZE, p->jointcapability), + ast_getformatname_multiple(s1, SIPBUFSIZE, p->jointcaps), ast_getformatname_multiple(s2, SIPBUFSIZE, p->owner->nativeformats)); } - p->owner->nativeformats = ast_codec_choose(&p->prefs, p->jointcapability, 1) | (p->capability & vpeercapability) | (p->capability & tpeercapability); - ast_set_read_format(p->owner, p->owner->readformat); - ast_set_write_format(p->owner, p->owner->writeformat); + + ast_codec_choose(&p->prefs, p->jointcaps, 1, &tmp_fmt); + + ast_format_cap_set(p->owner->nativeformats, &tmp_fmt); + ast_format_cap_joint_copy(p->caps, vpeercapability, p->owner->nativeformats); + ast_format_cap_joint_copy(p->caps, tpeercapability, p->owner->nativeformats); + + ast_set_read_format(p->owner, &p->owner->readformat); + ast_set_write_format(p->owner, &p->owner->writeformat); } if (ast_test_flag(&p->flags[1], SIP_PAGE2_CALL_ONHOLD) && (!ast_sockaddr_isnull(sa) || !ast_sockaddr_isnull(vsa) || !ast_sockaddr_isnull(tsa) || !ast_sockaddr_isnull(isa)) && (!sendonly || sendonly == -1)) { @@ -8846,8 +8905,14 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action ast_queue_frame(p->owner, &ast_null_frame); change_hold_state(p, req, TRUE, sendonly); } - - return 0; + +process_sdp_cleanup: + 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); + return res; } static int process_sdp_o(const char *o, struct sip_pvt *p) @@ -9001,10 +9066,10 @@ static int process_sdp_a_audio(const char *a, struct sip_pvt *p, struct ast_rtp_ 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 || !format.code) /* non-codec or not found */ + if (!format.asterisk_format) /* non-codec or not found */ continue; - ast_debug(1, "Setting framing for %s to %ld\n", ast_getformatname(format.code), framing); - ast_codec_pref_setsize(pref, format.code, framing); + 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); } @@ -9032,10 +9097,10 @@ static int process_sdp_a_audio(const char *a, struct sip_pvt *p, struct ast_rtp_ struct ast_rtp_payload_type payload; payload = ast_rtp_codecs_payload_lookup(newaudiortp, codec); - if (payload.code && payload.asterisk_format) { + if (payload.format.id && payload.asterisk_format) { unsigned int bit_rate; - switch (payload.code) { + switch ((int) payload.format.id) { case AST_FORMAT_SIREN7: if (sscanf(fmtp_string, "bitrate=%30u", &bit_rate) == 1) { if (bit_rate != 32000) { @@ -10373,31 +10438,34 @@ static int add_vidupdate(struct sip_request *req) } /*! \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, format_t codec, - struct ast_str **m_buf, struct ast_str **a_buf, - int debug, int *min_packet_size) +static void add_codec_to_sdp(const struct sip_pvt *p, + struct ast_format *format, + struct ast_str **m_buf, + struct ast_str **a_buf, + int debug, + int *min_packet_size) { int rtp_code; struct ast_format_list fmt; if (debug) - ast_verbose("Adding codec 0x%" PRIx64 " (%s) to SDP\n", codec, ast_getformatname(codec)); - if ((rtp_code = ast_rtp_codecs_payload_code(ast_rtp_instance_get_codecs(p->rtp), 1, codec)) == -1) + ast_verbose("Adding codec %d (%s) to SDP\n", format->id, ast_getformatname(format)); + if ((rtp_code = ast_rtp_codecs_payload_code(ast_rtp_instance_get_codecs(p->rtp), 1, format, 0)) == -1) return; if (p->rtp) { struct ast_codec_pref *pref = &ast_rtp_instance_get_codecs(p->rtp)->pref; - fmt = ast_codec_pref_getsize(pref, codec); + 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); - ast_str_append(a_buf, 0, "a=rtpmap:%d %s/%d\r\n", rtp_code, - ast_rtp_lookup_mime_subtype2(1, codec, - ast_test_flag(&p->flags[0], SIP_G726_NONSTANDARD) ? AST_RTP_OPT_G726_NONSTANDARD : 0), - ast_rtp_lookup_sample_rate2(1, codec)); + ast_str_append(a_buf, 0, "a=rtpmap:%d %s/%d\r\n", + rtp_code, + ast_rtp_lookup_mime_subtype2(1, format, 0, ast_test_flag(&p->flags[0], SIP_G726_NONSTANDARD) ? AST_RTP_OPT_G726_NONSTANDARD : 0), + ast_rtp_lookup_sample_rate2(1, format, 0)); - switch (codec) { + switch ((int) format->id) { case AST_FORMAT_G729A: /* 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); @@ -10434,7 +10502,7 @@ static void add_codec_to_sdp(const struct sip_pvt *p, format_t codec, /*! \brief Add video codec offer to SDP offer/answer body in INVITE or 200 OK */ /* This is different to the audio one now so we can add more caps later */ -static void add_vcodec_to_sdp(const struct sip_pvt *p, format_t codec, +static void add_vcodec_to_sdp(const struct sip_pvt *p, struct ast_format *format, struct ast_str **m_buf, struct ast_str **a_buf, int debug, int *min_packet_size) { @@ -10444,20 +10512,20 @@ static void add_vcodec_to_sdp(const struct sip_pvt *p, format_t codec, return; if (debug) - ast_verbose("Adding video codec 0x%" PRIx64 " (%s) to SDP\n", codec, ast_getformatname(codec)); + ast_verbose("Adding video codec %d (%s) to SDP\n", format->id, ast_getformatname(format)); - if ((rtp_code = ast_rtp_codecs_payload_code(ast_rtp_instance_get_codecs(p->vrtp), 1, codec)) == -1) + if ((rtp_code = ast_rtp_codecs_payload_code(ast_rtp_instance_get_codecs(p->vrtp), 1, format, 0)) == -1) return; ast_str_append(m_buf, 0, " %d", rtp_code); ast_str_append(a_buf, 0, "a=rtpmap:%d %s/%d\r\n", rtp_code, - ast_rtp_lookup_mime_subtype2(1, codec, 0), - ast_rtp_lookup_sample_rate2(1, codec)); + ast_rtp_lookup_mime_subtype2(1, format, 0, 0), + ast_rtp_lookup_sample_rate2(1, format, 0)); /* Add fmtp code here */ } /*! \brief Add text codec offer to SDP offer/answer body in INVITE or 200 OK */ -static void add_tcodec_to_sdp(const struct sip_pvt *p, int codec, +static void add_tcodec_to_sdp(const struct sip_pvt *p, struct ast_format *format, struct ast_str **m_buf, struct ast_str **a_buf, int debug, int *min_packet_size) { @@ -10467,19 +10535,20 @@ static void add_tcodec_to_sdp(const struct sip_pvt *p, int codec, return; if (debug) - ast_verbose("Adding text codec 0x%x (%s) to SDP\n", codec, ast_getformatname(codec)); + ast_verbose("Adding text codec %d (%s) to SDP\n", format->id, ast_getformatname(format)); - if ((rtp_code = ast_rtp_codecs_payload_code(ast_rtp_instance_get_codecs(p->trtp), 1, codec)) == -1) + if ((rtp_code = ast_rtp_codecs_payload_code(ast_rtp_instance_get_codecs(p->trtp), 1, format, 0)) == -1) return; ast_str_append(m_buf, 0, " %d", rtp_code); ast_str_append(a_buf, 0, "a=rtpmap:%d %s/%d\r\n", rtp_code, - ast_rtp_lookup_mime_subtype2(1, codec, 0), - ast_rtp_lookup_sample_rate2(1, codec)); + ast_rtp_lookup_mime_subtype2(1, format, 0, 0), + ast_rtp_lookup_sample_rate2(1, format, 0)); /* Add fmtp code here */ - if (codec == AST_FORMAT_T140RED) { - int t140code = ast_rtp_codecs_payload_code(ast_rtp_instance_get_codecs(p->trtp), 1, AST_FORMAT_T140); + 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); ast_str_append(a_buf, 0, "a=fmtp:%d %d/%d/%d\r\n", rtp_code, t140code, t140code, @@ -10518,14 +10587,14 @@ static void add_noncodec_to_sdp(const struct sip_pvt *p, int format, int rtp_code; if (debug) - ast_verbose("Adding non-codec 0x%x (%s) to SDP\n", format, ast_rtp_lookup_mime_subtype2(0, format, 0)); - if ((rtp_code = ast_rtp_codecs_payload_code(ast_rtp_instance_get_codecs(p->rtp), 0, format)) == -1) + ast_verbose("Adding non-codec 0x%x (%s) to SDP\n", format, ast_rtp_lookup_mime_subtype2(0, NULL, format, 0)); + if ((rtp_code = ast_rtp_codecs_payload_code(ast_rtp_instance_get_codecs(p->rtp), 0, NULL, format)) == -1) return; ast_str_append(m_buf, 0, " %d", rtp_code); ast_str_append(a_buf, 0, "a=rtpmap:%d %s/%d\r\n", rtp_code, - ast_rtp_lookup_mime_subtype2(0, format, 0), - ast_rtp_lookup_sample_rate2(0, format)); + ast_rtp_lookup_mime_subtype2(0, NULL, format, 0), + ast_rtp_lookup_sample_rate2(0, NULL, format)); if (format == AST_RTP_DTMF) /* Indicate we support DTMF and FLASH... */ ast_str_append(a_buf, 0, "a=fmtp:%d 0-16\r\n", rtp_code); } @@ -10651,9 +10720,10 @@ static void get_crypto_attrib(struct sip_srtp *srtp, const char **a_crypto) */ static enum sip_result add_sdp(struct sip_request *resp, struct sip_pvt *p, int oldsdp, int add_audio, int add_t38) { - format_t alreadysent = 0; + struct ast_format_cap *alreadysent = ast_format_cap_alloc_nolock(); + struct ast_format_cap *tmpcap = ast_format_cap_alloc_nolock(); + int res = AST_SUCCESS; int doing_directmedia = FALSE; - struct ast_sockaddr addr = { {0,} }; struct ast_sockaddr vaddr = { {0,} }; struct ast_sockaddr taddr = { {0,} }; @@ -10683,8 +10753,8 @@ static enum sip_result add_sdp(struct sip_request *resp, struct sip_pvt *p, int const char *v_a_crypto = NULL; const char *t_a_crypto = NULL; - format_t x; - format_t capability = 0; + int x; + struct ast_format tmp_fmt; int needaudio = FALSE; int needvideo = FALSE; int needtext = FALSE; @@ -10700,9 +10770,15 @@ static enum sip_result add_sdp(struct sip_request *resp, struct sip_pvt *p, int /* Set the SDP session name */ snprintf(subject, sizeof(subject), "s=%s\r\n", ast_strlen_zero(global_sdpsession) ? "-" : global_sdpsession); + if (!alreadysent || !tmpcap) { + res = AST_FAILURE; + goto add_sdp_cleanup; + } if (!p->rtp) { ast_log(LOG_WARNING, "No way to add SDP without an RTP structure\n"); - return AST_FAILURE; + res = AST_FAILURE; + goto add_sdp_cleanup; + } /* XXX We should not change properties in the SIP dialog until we have acceptance of the offer if this is a re-invite */ @@ -10717,9 +10793,9 @@ 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) && p->redircodecs) ? TRUE : FALSE; + doing_directmedia = (!ast_sockaddr_isnull(&p->redirip) && !(ast_format_cap_is_empty(p->redircaps))) ? TRUE : FALSE; /* Check if we need video in this call */ - if ((p->jointcapability & AST_FORMAT_VIDEO_MASK) && !p->novideo) { + if ((ast_format_cap_has_type(p->jointcaps, AST_FORMAT_TYPE_VIDEO)) && !p->novideo) { if (p->vrtp) { needvideo = TRUE; ast_debug(2, "This call needs video offers!\n"); @@ -10727,7 +10803,7 @@ 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 ((p->jointcapability & AST_FORMAT_TEXT_MASK) && !p->notext) { + if ((ast_format_cap_has_type(p->jointcaps, AST_FORMAT_TYPE_TEXT)) && !p->notext) { if (sipdebug_text) ast_verbose("We think we can do text\n"); if (p->trtp) { @@ -10767,20 +10843,20 @@ static enum sip_result add_sdp(struct sip_request *resp, struct sip_pvt *p, int hold = "a=sendrecv\r\n"; } - capability = p->jointcapability; + 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), capability), + 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->prefcodec)); + ast_debug(1, "** Our prefcodec: %s \n", ast_getformatname_multiple(codecbuf, sizeof(codecbuf), p->prefcaps)); if (doing_directmedia) { - capability &= p->redircodecs; - ast_debug(1, "** Our native-bridge filtered capablity: %s\n", ast_getformatname_multiple(codecbuf, sizeof(codecbuf), capability)); + 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 (capability & AST_FORMAT_AUDIO_MASK) + if (ast_format_cap_has_type(tmpcap, AST_FORMAT_TYPE_AUDIO)) needaudio = TRUE; if (debug) { @@ -10833,45 +10909,55 @@ static enum sip_result add_sdp(struct sip_request *resp, struct sip_pvt *p, int /* 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) { - format_t codec = p->prefcodec & AST_FORMAT_AUDIO_MASK; - - add_codec_to_sdp(p, codec, &m_audio, &a_audio, debug, &min_audio_packet_size); - alreadysent |= codec; + if (ast_format_cap_has_joint(tmpcap, p->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) { + continue; + } + add_codec_to_sdp(p, &tmp_fmt, &m_audio, &a_audio, debug, &min_audio_packet_size); + ast_format_cap_add(alreadysent, &tmp_fmt); + } + ast_format_cap_iter_end(p->prefcaps); } /* Start by sending our preferred audio/video codecs */ - for (x = 0; x < 64; x++) { - format_t codec; - - if (!(codec = ast_codec_pref_index(&p->prefs, x))) + for (x = 0; x < AST_CODEC_PREF_SIZE; x++) { + if (!(ast_codec_pref_index(&p->prefs, x, &tmp_fmt))) break; - if (!(capability & codec)) + if (!(ast_format_cap_iscompatible(tmpcap, &tmp_fmt))) continue; - if (alreadysent & codec) + if (ast_format_cap_iscompatible(alreadysent, &tmp_fmt)) continue; - add_codec_to_sdp(p, codec, &m_audio, &a_audio, debug, &min_audio_packet_size); - alreadysent |= codec; + 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); + } 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); + } + + ast_format_cap_add(alreadysent, &tmp_fmt); } /* Now send any other common audio and video codecs, and non-codec formats: */ - for (x = 1ULL; x <= (needtext ? AST_FORMAT_TEXT_MASK : (needvideo ? AST_FORMAT_VIDEO_MASK : AST_FORMAT_AUDIO_MASK)); x <<= 1) { - if (!(capability & x)) /* Codec not requested */ + ast_format_cap_iter_start(tmpcap); + while (!(ast_format_cap_iter_next(tmpcap, &tmp_fmt))) { + if (ast_format_cap_iscompatible(alreadysent, &tmp_fmt)) continue; - if (alreadysent & x) /* Already added to SDP */ - continue; - - if (x & AST_FORMAT_AUDIO_MASK) - add_codec_to_sdp(p, x, &m_audio, &a_audio, debug, &min_audio_packet_size); - else if (x & AST_FORMAT_VIDEO_MASK) - add_vcodec_to_sdp(p, x, &m_video, &a_video, debug, &min_video_packet_size); - else if (x & AST_FORMAT_TEXT_MASK) - add_tcodec_to_sdp(p, x, &m_text, &a_text, debug, &min_text_packet_size); + 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); + } 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); + } } + ast_format_cap_iter_end(tmpcap); /* Now add DTMF RFC2833 telephony-event as a codec */ for (x = 1LL; x <= AST_RTP_MAX; x <<= 1) { @@ -11024,9 +11110,13 @@ static enum sip_result add_sdp(struct sip_request *resp, struct sip_pvt *p, int ao2_t_unlink(dialogs_rtpcheck, p, "unlink pvt into dialogs_rtpcheck container"); ao2_t_link(dialogs_rtpcheck, p, "link pvt into dialogs_rtpcheck container"); - ast_debug(3, "Done building SDP. Settling with this capability: %s\n", ast_getformatname_multiple(buf, SIPBUFSIZE, capability)); + ast_debug(3, "Done building SDP. Settling with this capability: %s\n", ast_getformatname_multiple(buf, SIPBUFSIZE, tmpcap)); - return AST_SUCCESS; +add_sdp_cleanup: + alreadysent = ast_format_cap_destroy(alreadysent); + tmpcap = ast_format_cap_destroy(tmpcap); + + return res; } /*! \brief Used for 200 OK and 183 early media */ @@ -15257,14 +15347,21 @@ static enum check_auth_result check_peer_ok(struct sip_pvt *p, char *of, p->amaflags = peer->amaflags; p->callgroup = peer->callgroup; p->pickupgroup = peer->pickupgroup; - p->capability = peer->capability; + ast_format_cap_copy(p->caps, peer->caps); + ast_format_cap_copy(p->jointcaps, peer->caps); p->prefs = peer->prefs; - p->jointcapability = peer->capability; if (peer->maxforwards > 0) { p->maxforwards = peer->maxforwards; } - if (p->peercapability) - p->jointcapability &= p->peercapability; + 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); + } + } p->maxcallbitrate = peer->maxcallbitrate; if ((ast_test_flag(&p->flags[0], SIP_DTMF) == SIP_DTMF_RFC2833) || (ast_test_flag(&p->flags[0], SIP_DTMF) == SIP_DTMF_AUTO)) @@ -16333,15 +16430,15 @@ static char *sip_prune_realtime(struct ast_cli_entry *e, int cmd, struct ast_cli static void print_codec_to_cli(int fd, struct ast_codec_pref *pref) { int x; - format_t codec; + struct ast_format codec; - for(x = 0; x < 64 ; x++) { - codec = ast_codec_pref_index(pref, x); - if (!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, "%s", ast_getformatname(&codec)); ast_cli(fd, ":%d", pref->framing[x]); - if (x < 31 && ast_codec_pref_index(pref, x + 1)) + if (x < 31 && ast_codec_pref_index(pref, x + 1, &codec)) ast_cli(fd, ","); } if (!x) @@ -16529,7 +16626,7 @@ static char *_sip_show_peer(int type, int fd, struct mansession *s, const struct struct ast_variable *v; struct sip_auth *auth; int x = 0, load_realtime; - format_t codec = 0; + struct ast_format codec; int realtimepeers; realtimepeers = ast_check_realtime("sippeers"); @@ -16644,7 +16741,7 @@ static char *_sip_show_peer(int type, int fd, struct mansession *s, const struct ast_cli(fd, "\n"); ast_cli(fd, " Codecs : "); - ast_getformatname_multiple(codec_buf, sizeof(codec_buf) -1, peer->capability); + 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); @@ -16738,16 +16835,16 @@ static char *_sip_show_peer(int type, int fd, struct mansession *s, const struct 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->capability); + 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 < 64 ; x++) { - codec = ast_codec_pref_index(pref, x); - if (!codec) + 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 < 63 && ast_codec_pref_index(pref, x+1)) + } + 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, ","); } @@ -17243,7 +17340,7 @@ 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.capability); + 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); @@ -17400,7 +17497,7 @@ static int show_channels_cb(void *__cur, void *__arg, int flags) ast_cli(arg->fd, FORMAT, ast_sockaddr_stringify_addr(dst), S_OR(cur->username, S_OR(cur->cid_num, "(None)")), cur->callid, - ast_getformatname_multiple(formatbuf, sizeof(formatbuf), cur->owner ? cur->owner->nativeformats : 0), + cur->owner ? ast_getformatname_multiple(formatbuf, sizeof(formatbuf), cur->owner->nativeformats) : "(nothing)", AST_CLI_YESNO(ast_test_flag(&cur->flags[1], SIP_PAGE2_CALL_ONHOLD)), cur->needdestroy ? "(d)" : "", cur->lastmsg , @@ -17649,11 +17746,11 @@ 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 ? cur->owner->name : "<none>"); - ast_cli(a->fd, " Our Codec Capability: %s\n", ast_getformatname_multiple(formatbuf, sizeof(formatbuf), cur->capability)); + ast_cli(a->fd, " Our Codec Capability: %s\n", ast_getformatname_multiple(formatbuf, sizeof(formatbuf), cur->caps)); 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->peercapability)); - ast_cli(a->fd, " Joint Codec Capability: %s\n", ast_getformatname_multiple(formatbuf, sizeof(formatbuf), cur->jointcapability)); - ast_cli(a->fd, " Format: %s\n", ast_getformatname_multiple(formatbuf, sizeof(formatbuf), cur->owner ? cur->owner->nativeformats : 0) ); + 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), cur->owner->nativeformats) : "(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); @@ -18558,13 +18655,13 @@ 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->capability); + ast_getformatname_multiple(buf, len -1, peer->caps); } else if (!strcasecmp(colname, "encryption")) { snprintf(buf, len, "%d", ast_test_flag(&peer->flags[1], SIP_PAGE2_USE_SRTP)); } else if (!strncasecmp(colname, "chanvar[", 8)) { char *chanvar=colname + 8; struct ast_variable *v; - + chanvar = strsep(&chanvar, "]"); for (v = peer->chanvars ; v ; v = v->next) { if (!strcasecmp(v->name, chanvar)) { @@ -18573,12 +18670,12 @@ static int function_sippeer(struct ast_channel *chan, const char *cmd, char *dat } } else if (!strncasecmp(colname, "codec[", 6)) { char *codecnum; - format_t codec = 0; - + struct ast_format codec; + codecnum = colname + 6; /* move past the '[' */ codecnum = strsep(&codecnum, "]"); /* trim trailing ']' if any */ - if((codec = ast_codec_pref_index(&peer->prefs, atoi(codecnum)))) { - ast_copy_string(buf, ast_getformatname(codec), len); + if((ast_codec_pref_index(&peer->prefs, atoi(codecnum), &codec))) { + ast_copy_string(buf, ast_getformatname(&codec), len); } else { buf[0] = '\0'; } @@ -21484,7 +21581,7 @@ static int handle_request_invite(struct sip_pvt *p, struct sip_request *req, int } ast_queue_control(p->owner, AST_CONTROL_SRCUPDATE); } else { - p->jointcapability = p->capability; + ast_format_cap_copy(p->jointcaps, p->caps); 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" @@ -21566,7 +21663,7 @@ static int handle_request_invite(struct sip_pvt *p, struct sip_request *req, int goto request_invite_cleanup; } } else { /* No SDP in invite, call control session */ - p->jointcapability = p->capability; + ast_format_cap_copy(p->jointcaps, p->caps); ast_debug(2, "No SDP in Invite, third party call control\n"); } @@ -21968,7 +22065,7 @@ static int handle_request_invite(struct sip_pvt *p, struct sip_request *req, int if (p && (p->autokillid == -1)) { const char *msg; - if (!p->jointcapability) + if ((ast_format_cap_is_empty(p->jointcaps))) msg = "488 Not Acceptable Here (codec error)"; else { ast_log(LOG_NOTICE, "Unable to create/find SIP channel for this INVITE\n"); @@ -25309,12 +25406,13 @@ static int sip_devicestate(void *data) * or SIP/host!dnid * \endverbatim */ -static struct ast_channel *sip_request_call(const char *type, format_t format, const struct ast_channel *requestor, void *data, int *cause) +static struct ast_channel *sip_request_call(const char *type, struct ast_format_cap *cap, const struct ast_channel *requestor, void *data, int *cause) { struct sip_pvt *p; struct ast_channel *tmpc = NULL; char *ext = NULL, *host; char tmp[256]; + char tmp2[256]; char *dest = data; char *dnid; char *secret = NULL; @@ -25325,7 +25423,6 @@ static struct ast_channel *sip_request_call(const char *type, format_t format, c char *remote_address; enum sip_transport transport = 0; struct ast_sockaddr remote_address_sa = { {0,} }; - format_t oldformat = format; AST_DECLARE_APP_ARGS(args, AST_APP_ARG(peerorhost); AST_APP_ARG(exten); @@ -25339,13 +25436,14 @@ static struct ast_channel *sip_request_call(const char *type, format_t format, c * configured from sip.conf, and sip_tech.capabilities, which is * hardwired to all audio formats. */ - format &= AST_FORMAT_AUDIO_MASK; - if (!format) { - ast_log(LOG_NOTICE, "Asked to get a channel of unsupported format %s while capability is %s\n", ast_getformatname(oldformat), ast_getformatname(sip_cfg.capability)); + if (!(ast_format_cap_has_type(cap, AST_FORMAT_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)); *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), oldformat)); + ast_debug(1, "Asked to create a SIP channel with formats: %s\n", ast_getformatname_multiple(tmp, sizeof(tmp), cap)); if (ast_strlen_zero(dest)) { ast_log(LOG_ERROR, "Unable to create channel with empty destination.\n"); @@ -25488,8 +25586,9 @@ static struct ast_channel *sip_request_call(const char *type, format_t format, c #if 0 printf("Setting up to call extension '%s' at '%s'\n", ext ? ext : "<none>", host); #endif - p->prefcodec = oldformat; /* Format for this call */ - p->jointcapability = oldformat & p->capability; + ast_format_cap_append(p->prefcaps, cap); + ast_format_cap_joint_copy(cap, p->caps, p->jointcaps); + sip_pvt_lock(p); tmpc = sip_new(p, AST_STATE_DOWN, host, requestor ? requestor->linkedid : NULL); /* Place the call */ if (sip_cfg.callevents) @@ -25915,7 +26014,7 @@ 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); - peer->capability = sip_cfg.capability; + ast_format_cap_copy(peer->caps, sip_cfg.caps); peer->maxcallbitrate = default_maxcallbitrate; peer->rtptimeout = global_rtptimeout; peer->rtpholdtimeout = global_rtpholdtimeout; @@ -26063,9 +26162,13 @@ static struct sip_peer *build_peer(const char *name, struct ast_variable *v, str if (!(peer->the_mark)) firstpass = 0; } else { - if (!(peer = ao2_t_alloc(sizeof(*peer), sip_destroy_peer_fn, "allocate a peer struct"))) + if (!(peer = ao2_t_alloc(sizeof(*peer), sip_destroy_peer_fn, "allocate a peer struct"))) { return NULL; - + } + if (!(peer->caps = ast_format_cap_alloc_nolock())) { + ao2_t_ref(peer, -1, "failed to allocate format capabilities, drop peer"); + return NULL; + } if (ast_string_field_init(peer, 512)) { ao2_t_ref(peer, -1, "failed to string_field_init, drop peer"); return NULL; @@ -26356,12 +26459,12 @@ static struct sip_peer *build_peer(const char *name, struct ast_variable *v, str } else if (!strcasecmp(v->name, "pickupgroup")) { peer->pickupgroup = ast_get_group(v->value); } else if (!strcasecmp(v->name, "allow")) { - int error = ast_parse_allow_disallow(&peer->prefs, &peer->capability, v->value, TRUE); + int error = ast_parse_allow_disallow(&peer->prefs, 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->capability, v->value, FALSE); + int error = ast_parse_allow_disallow(&peer->prefs, 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); } @@ -26697,6 +26800,22 @@ static int peer_markall_func(void *device, void *arg, int flags) return 0; } +/*! + * \internal + * \brief If no default formats are set in config, these are used + */ +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)); +} + /*! \brief Re-read SIP.conf config file \note This function reloads all config data, except for active peers (with registrations). They will only @@ -26852,7 +26971,7 @@ static int reload_config(enum channelreloadreason reason) /* Reset channel settings to default before re-configuring */ sip_cfg.allow_external_domains = DEFAULT_ALLOW_EXT_DOM; /* Allow external invites */ sip_cfg.regcontext[0] = '\0'; - sip_cfg.capability = DEFAULT_CAPABILITY; + sip_set_default_format_capabilities(sip_cfg.caps); sip_cfg.regextenonqualify = DEFAULT_REGEXTENONQUALIFY; sip_cfg.notifyringing = DEFAULT_NOTIFYRINGING; sip_cfg.notifycid = DEFAULT_NOTIFYCID; @@ -27258,12 +27377,12 @@ 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.capability, v->value, TRUE); + int error = ast_parse_allow_disallow(&default_prefs, 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.capability, v->value, FALSE); + int error = ast_parse_allow_disallow(&default_prefs, 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); } @@ -27918,7 +28037,7 @@ static enum ast_rtp_glue_result sip_get_trtp_peer(struct ast_channel *chan, stru return res; } -static int sip_set_rtp_peer(struct ast_channel *chan, struct ast_rtp_instance *instance, struct ast_rtp_instance *vinstance, struct ast_rtp_instance *tinstance, format_t codecs, int nat_active) +static int sip_set_rtp_peer(struct ast_channel *chan, struct ast_rtp_instance *instance, struct ast_rtp_instance *vinstance, struct ast_rtp_instance *tinstance, const struct ast_format_cap *cap, int nat_active) { struct sip_pvt *p; int changed = 0; @@ -27966,8 +28085,8 @@ 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 (codecs && (p->redircodecs != codecs)) { - p->redircodecs = codecs; + if (cap && !(ast_format_cap_is_empty(cap) && !(ast_format_cap_identical(p->redircaps, cap)))) { + ast_format_cap_copy(p->redircaps, cap); changed = 1; } if (changed && !ast_test_flag(&p->flags[0], SIP_GOTREFER) && !ast_test_flag(&p->flags[0], SIP_DEFER_BYE_ON_TRANSFER)) { @@ -27990,10 +28109,10 @@ static int sip_set_rtp_peer(struct ast_channel *chan, struct ast_rtp_instance *i return 0; } -static format_t sip_get_codec(struct ast_channel *chan) +static void sip_get_codec(struct ast_channel *chan, struct ast_format_cap *result) { struct sip_pvt *p = chan->tech_pvt; - return p->peercapability ? p->peercapability : p->capability; + ast_format_cap_append(result, ast_format_cap_is_empty(p->peercaps) ? p->caps : p->peercaps); } static struct ast_rtp_glue sip_rtp_glue = { @@ -28955,7 +29074,7 @@ static int peers_data_provider_get(const struct ast_data_search *search, ast_data_add_str(enum_node, "text", ast_describe_caller_presentation(peer->callingpres)); /* codecs */ - ast_data_add_codecs(data_peer, "codecs", peer->capability); + ast_data_add_codecs(data_peer, "codecs", peer->caps); if (!ast_data_search_match(search, data_peer)) { ast_data_remove_node(data_root, data_peer); @@ -28982,6 +29101,10 @@ static const struct ast_data_entry sip_data_providers[] = { static int load_module(void) { ast_verbose("SIP channel loading...\n"); + + if (!(sip_tech.capabilities = ast_format_cap_alloc())) { + return AST_MODULE_LOAD_FAILURE; + } /* the fact that ao2_containers can't resize automatically is a major worry! */ /* if the number of objects gets above MAX_XXX_BUCKETS, things will slow down */ peers = ao2_t_container_alloc(HASH_PEER_SIZE, peer_hash_cb, peer_cmp_cb, "allocate peers"); @@ -28990,7 +29113,12 @@ static int load_module(void) dialogs_needdestroy = ao2_t_container_alloc(HASH_DIALOG_SIZE, dialog_hash_cb, dialog_cmp_cb, "allocate dialogs_needdestroy"); dialogs_rtpcheck = ao2_t_container_alloc(HASH_DIALOG_SIZE, dialog_hash_cb, dialog_cmp_cb, "allocate dialogs for rtpchecks"); threadt = ao2_t_container_alloc(HASH_DIALOG_SIZE, threadt_hash_cb, threadt_cmp_cb, "allocate threadt table"); - + + if (!(sip_cfg.caps = ast_format_cap_alloc())) { + return AST_MODULE_LOAD_FAILURE; + } + ast_format_cap_add_all_by_type(sip_tech.capabilities, AST_FORMAT_TYPE_AUDIO); + ASTOBJ_CONTAINER_INIT(®l); /* Registry object list -- not searched for anything */ ASTOBJ_CONTAINER_INIT(&submwil); /* MWI subscription object list */ @@ -29257,6 +29385,9 @@ static int unload_module(void) sip_reqresp_parser_exit(); sip_unregister_tests(); + ast_format_cap_destroy(sip_tech.capabilities); + sip_cfg.caps = ast_format_cap_destroy(sip_cfg.caps); + return 0; } |