From: Kinsey Moore Date: Thu, 10 Nov 2011 18:14:20 +0000 (+0000) Subject: Fix several bugs with SDP parsing and well-formedness of responses X-Git-Tag: 10.1.0-rc1~103 X-Git-Url: http://git.ipfire.org/cgi-bin/gitweb.cgi?a=commitdiff_plain;h=f20fd93eff3593b3ad3bbb697068d2271cfa6765;p=thirdparty%2Fasterisk.git Fix several bugs with SDP parsing and well-formedness of responses Fix bug ASTERISK-16558 which dealt with the order of responses to incoming streams defined by SDP. Fix unreported bug where offering multiple same-type streams would cause Asterisk to reply with an incorrect SDP response missing one or more streams without a proper declination. Fix bugs related to a single non-audio stream being offered with responses requesting codecs that were not offered in the initial invite along with an additional audio stream that was not in the initial invite. Review: https://reviewboard.asterisk.org/r/1516/ ........ Merged revisions 344385 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@344386 65c4cc65-6c06-0410-ace0-fbb531ad65f3 --- diff --git a/channels/chan_sip.c b/channels/chan_sip.c index 39b6fbfa97..250bd0689c 100644 --- a/channels/chan_sip.c +++ b/channels/chan_sip.c @@ -9010,9 +9010,13 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action ast_log(LOG_WARNING, "unknown SDP media protocol in offer: %s\n", protocol); continue; } + if (p->offered_media[SDP_AUDIO].order_offered) { + ast_log(LOG_WARNING, "Multiple audio streams are not supported\n"); + res = -3; + goto process_sdp_cleanup; + } audio = TRUE; - p->offered_media[SDP_AUDIO].offered = TRUE; - numberofmediastreams++; + p->offered_media[SDP_AUDIO].order_offered = ++numberofmediastreams; portno = x; /* Scan through the RTP payload types specified in a "m=" line: */ @@ -9038,10 +9042,14 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action ast_log(LOG_WARNING, "unknown SDP media protocol in offer: %s\n", protocol); continue; } + if (p->offered_media[SDP_VIDEO].order_offered) { + ast_log(LOG_WARNING, "Multiple video streams are not supported\n"); + res = -3; + goto process_sdp_cleanup; + } video = TRUE; p->novideo = FALSE; - p->offered_media[SDP_VIDEO].offered = TRUE; - numberofmediastreams++; + p->offered_media[SDP_VIDEO].order_offered = ++numberofmediastreams; vportno = x; /* Scan through the RTP payload types specified in a "m=" line: */ @@ -9060,10 +9068,14 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action /* Search for text media definition */ } else if ((sscanf(m, "text %30u/%30u RTP/AVP %n", &x, &numberofports, &len) == 2 && len > 0 && x) || (sscanf(m, "text %30u RTP/AVP %n", &x, &len) == 1 && len > 0 && x)) { + if (p->offered_media[SDP_TEXT].order_offered) { + ast_log(LOG_WARNING, "Multiple text streams are not supported\n"); + res = -3; + goto process_sdp_cleanup; + } text = TRUE; p->notext = FALSE; - p->offered_media[SDP_TEXT].offered = TRUE; - numberofmediastreams++; + p->offered_media[SDP_TEXT].order_offered = ++numberofmediastreams; tportno = x; /* Scan through the RTP payload types specified in a "m=" line: */ @@ -9082,12 +9094,16 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action /* Search for image media definition */ } else if (p->udptl && ((sscanf(m, "image %30u udptl t38%n", &x, &len) == 1 && len > 0 && x) || (sscanf(m, "image %30u UDPTL t38%n", &x, &len) == 1 && len > 0 && x) )) { + if (p->offered_media[SDP_IMAGE].order_offered) { + ast_log(LOG_WARNING, "Multiple T.38 streams are not supported\n"); + res = -3; + goto process_sdp_cleanup; + } image = TRUE; if (debug) ast_verbose("Got T.38 offer in SDP in dialog %s\n", p->callid); - p->offered_media[SDP_IMAGE].offered = TRUE; + p->offered_media[SDP_IMAGE].order_offered = ++numberofmediastreams; udptlportno = x; - numberofmediastreams++; if (p->t38.state != T38_ENABLED) { memset(&p->t38.their_parms, 0, sizeof(p->t38.their_parms)); @@ -9190,13 +9206,6 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action 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"); - 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"); res = -4; @@ -9241,7 +9250,7 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action ast_format_cap_append(newpeercapability, tpeercapability); ast_format_cap_joint_copy(p->caps, newpeercapability, newjointcapability); - if (ast_format_cap_is_empty(newjointcapability) && (portno != -1)) { + if (ast_format_cap_is_empty(newjointcapability) && udptlportno == -1) { ast_log(LOG_NOTICE, "No compatible codecs, not accepting this offer!\n"); /* Do NOT Change current setting */ res = -1; @@ -9272,6 +9281,17 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action ast_rtp_lookup_mime_multiple2(s3, NULL, newnoncodeccapability, 0, 0)); } + /* We are now ready to change the sip session and p->rtp and p->vrtp 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 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 */ + ast_codec_choose(&p->prefs, p->jointcaps, 1, &tmp_fmt); + ast_format_cap_set(p->jointcaps, &tmp_fmt); + } + /* Setup audio address and port */ if (p->rtp) { if (portno > 0) { @@ -9281,16 +9301,6 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action ast_verbose("Peer audio RTP is at port %s\n", ast_sockaddr_stringify(sa)); } - /* We are now ready to change the sip session and p->rtp and p->vrtp 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 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 */ - 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); /* Ensure RTCP is enabled since it may be inactive @@ -11670,47 +11680,94 @@ static enum sip_result add_sdp(struct sip_request *resp, struct sip_pvt *p, int add_content(resp, owner); add_content(resp, subject); add_content(resp, connection); - if (needvideo) /* only if video response is appropriate */ + /* only if video response is appropriate */ + if (needvideo) { add_content(resp, bandwidth); - add_content(resp, session_time); - if (needaudio) { - add_content(resp, m_audio->str); - add_content(resp, a_audio->str); - add_content(resp, hold); - if (a_crypto) { - add_content(resp, a_crypto); - } - } else if (p->offered_media[SDP_AUDIO].offered) { - snprintf(dummy_answer, sizeof(dummy_answer), "m=audio 0 RTP/AVP %s\r\n", p->offered_media[SDP_AUDIO].codecs); - add_content(resp, dummy_answer); - } - if (needvideo) { /* only if video response is appropriate */ - add_content(resp, m_video->str); - add_content(resp, a_video->str); - add_content(resp, hold); /* Repeat hold for the video stream */ - if (v_a_crypto) { - add_content(resp, v_a_crypto); - } - } else if (p->offered_media[SDP_VIDEO].offered) { - snprintf(dummy_answer, sizeof(dummy_answer), "m=video 0 RTP/AVP %s\r\n", p->offered_media[SDP_VIDEO].codecs); - add_content(resp, dummy_answer); - } - if (needtext) { /* only if text response is appropriate */ - add_content(resp, m_text->str); - add_content(resp, a_text->str); - add_content(resp, hold); /* Repeat hold for the text stream */ - if (t_a_crypto) { - add_content(resp, t_a_crypto); - } - } else if (p->offered_media[SDP_TEXT].offered) { - snprintf(dummy_answer, sizeof(dummy_answer), "m=text 0 RTP/AVP %s\r\n", p->offered_media[SDP_TEXT].codecs); - add_content(resp, dummy_answer); } - if (add_t38) { - add_content(resp, m_modem->str); - add_content(resp, a_modem->str); - } else if (p->offered_media[SDP_IMAGE].offered) { - add_content(resp, "m=image 0 udptl t38\r\n"); + add_content(resp, session_time); + /* if this is a response to an invite, order our offers properly */ + if (p->offered_media[SDP_AUDIO].order_offered || + p->offered_media[SDP_VIDEO].order_offered || + p->offered_media[SDP_TEXT].order_offered || + p->offered_media[SDP_IMAGE].order_offered) { + int i; + /* we have up to 3 streams as limited by process_sdp */ + for (i = 1; i <= 3; i++) { + if (p->offered_media[SDP_AUDIO].order_offered == i) { + if (needaudio) { + add_content(resp, m_audio->str); + add_content(resp, a_audio->str); + add_content(resp, hold); + if (a_crypto) { + add_content(resp, a_crypto); + } + } else { + snprintf(dummy_answer, sizeof(dummy_answer), "m=audio 0 RTP/AVP %s\r\n", p->offered_media[SDP_AUDIO].codecs); + add_content(resp, dummy_answer); + } + } else if (p->offered_media[SDP_VIDEO].order_offered == i) { + if (needvideo) { /* only if video response is appropriate */ + add_content(resp, m_video->str); + add_content(resp, a_video->str); + add_content(resp, hold); /* Repeat hold for the video stream */ + if (v_a_crypto) { + add_content(resp, v_a_crypto); + } + } else { + snprintf(dummy_answer, sizeof(dummy_answer), "m=video 0 RTP/AVP %s\r\n", p->offered_media[SDP_VIDEO].codecs); + add_content(resp, dummy_answer); + } + } else if (p->offered_media[SDP_TEXT].order_offered == i) { + if (needtext) { /* only if text response is appropriate */ + add_content(resp, m_text->str); + add_content(resp, a_text->str); + add_content(resp, hold); /* Repeat hold for the text stream */ + if (t_a_crypto) { + add_content(resp, t_a_crypto); + } + } else { + snprintf(dummy_answer, sizeof(dummy_answer), "m=text 0 RTP/AVP %s\r\n", p->offered_media[SDP_TEXT].codecs); + add_content(resp, dummy_answer); + } + } else if (p->offered_media[SDP_IMAGE].order_offered == i) { + if (add_t38) { + add_content(resp, m_modem->str); + add_content(resp, a_modem->str); + } else { + add_content(resp, "m=image 0 udptl t38\r\n"); + } + } + } + } else { + /* generate new SDP from scratch, no offers */ + if (needaudio) { + add_content(resp, m_audio->str); + add_content(resp, a_audio->str); + add_content(resp, hold); + if (a_crypto) { + add_content(resp, a_crypto); + } + } + if (needvideo) { /* only if video response is appropriate */ + add_content(resp, m_video->str); + add_content(resp, a_video->str); + add_content(resp, hold); /* Repeat hold for the video stream */ + if (v_a_crypto) { + add_content(resp, v_a_crypto); + } + } + if (needtext) { /* only if text response is appropriate */ + add_content(resp, m_text->str); + add_content(resp, a_text->str); + add_content(resp, hold); /* Repeat hold for the text stream */ + if (t_a_crypto) { + add_content(resp, t_a_crypto); + } + } + if (add_t38) { + add_content(resp, m_modem->str); + add_content(resp, a_modem->str); + } } /* Update lastrtprx when we send our SDP */ diff --git a/channels/sip/include/sip.h b/channels/sip/include/sip.h index 7a0a3bb577..36ed022353 100644 --- a/channels/sip/include/sip.h +++ b/channels/sip/include/sip.h @@ -943,7 +943,7 @@ struct sip_st_cfg { /*! \brief Structure for remembering offered media in an INVITE, to make sure we reply to all media streams. In theory. In practise, we try our best. */ struct offered_media { - int offered; + int order_offered; /*!< Order the media was offered in. Not offered is 0 */ char codecs[128]; };