Dashboard

chan_sip: Asterisk fails to re-activate an inactive media session when an

offer does not contain a=sendrecv

This test is to ensure that Asterisk correctly applies the direction of the

media stream when a=<sendonly|recvonly|inactive|sendrecv> is missing from the

offer's SDP. The expected behavior is for Asterisk to apply "sendrecv" as the

direction of the media stream when no direction attribute is present in an

offer's SDP.

According to RFC 4566 (Section 6. SDP Attributes): "If none of the attributes

"sendonly", "recvonly", "inactive", and "sendrecv" is present, "sendrecv"

SHOULD be assumed as the default for sessions that are not of the conference

type "broadcast" or "H332" [...]"

The test scenario:

1. From Phone A, send an offer to Phone B to establish a call

2. From Phone B, send an offer to Phone A to put the call on hold.

3. Observe that the MOH start event occurs.

4. From Phone B, send an offer to Phone A to 'un-hold' the call (ensure that

the direction attribute from the offer's SDP is omitted)

5. Observe that the MOH stop event occurs.

ASTERISK-24824 #close

Reported By: Ashley Sanders

Review: https://reviewboard.asterisk.org/r/4442/

........

Merged revisions 6483 from file:///srv/subversion/repos/testsuite/asterisk/trunk

chan_sip: Asterisk fails to re-activate an inactive media session when an

offer does not contain a=sendrecv

This test is to ensure that Asterisk correctly applies the direction of the

media stream when a=<sendonly|recvonly|inactive|sendrecv> is missing from the

offer's SDP. The expected behavior is for Asterisk to apply "sendrecv" as the

direction of the media stream when no direction attribute is present in an

offer's SDP.

According to RFC 4566 (Section 6. SDP Attributes): "If none of the attributes

"sendonly", "recvonly", "inactive", and "sendrecv" is present, "sendrecv"

SHOULD be assumed as the default for sessions that are not of the conference

type "broadcast" or "H332" [...]"

The test scenario:

1. From Phone A, send an offer to Phone B to establish a call

2. From Phone B, send an offer to Phone A to put the call on hold.

3. Observe that the MOH start event occurs.

4. From Phone B, send an offer to Phone A to 'un-hold' the call (ensure that

the direction attribute from the offer's SDP is omitted)

5. Observe that the MOH stop event occurs.

ASTERISK-24824 #close

Reported By: Ashley Sanders

Review: https://reviewboard.asterisk.org/r/4442/

........

Merged revisions 6483 from file:///srv/subversion/repos/testsuite/asterisk/trunk

chan_sip: Asterisk fails to re-activate an inactive media session when an

offer does not contain a=sendrecv

This test is to ensure that Asterisk correctly applies the direction of the

media stream when a=<sendonly|recvonly|inactive|sendrecv> is missing from the

offer's SDP. The expected behavior is for Asterisk to apply "sendrecv" as the

direction of the media stream when no direction attribute is present in an

offer's SDP.

According to RFC 4566 (Section 6. SDP Attributes): "If none of the attributes

"sendonly", "recvonly", "inactive", and "sendrecv" is present, "sendrecv"

SHOULD be assumed as the default for sessions that are not of the conference

type "broadcast" or "H332" [...]"

The test scenario:

1. From Phone A, send an offer to Phone B to establish a call

2. From Phone B, send an offer to Phone A to put the call on hold.

3. Observe that the MOH start event occurs.

4. From Phone B, send an offer to Phone A to 'un-hold' the call (ensure that

the direction attribute from the offer's SDP is omitted)

5. Observe that the MOH stop event occurs.

ASTERISK-24824 #close

Reported By: Ashley Sanders

Review: https://reviewboard.asterisk.org/r/4442/

........

Merged revisions 6483 from file:///srv/subversion/repos/testsuite/asterisk/trunk

chan_sip: Asterisk fails to re-activate an inactive media session when an

offer does not contain a=sendrecv

This test is to ensure that Asterisk correctly applies the direction of the

media stream when a=<sendonly|recvonly|inactive|sendrecv> is missing from the

offer's SDP. The expected behavior is for Asterisk to apply "sendrecv" as the

direction of the media stream when no direction attribute is present in an

offer's SDP.

According to RFC 4566 (Section 6. SDP Attributes): "If none of the attributes

"sendonly", "recvonly", "inactive", and "sendrecv" is present, "sendrecv"

SHOULD be assumed as the default for sessions that are not of the conference

type "broadcast" or "H332" [...]"

The test scenario:

1. From Phone A, send an offer to Phone B to establish a call

2. From Phone B, send an offer to Phone A to put the call on hold.

3. Observe that the MOH start event occurs.

4. From Phone B, send an offer to Phone A to 'un-hold' the call (ensure that

the direction attribute from the offer's SDP is omitted)

5. Observe that the MOH stop event occurs.

ASTERISK-24824 #close

Reported By: Ashley Sanders

Review: https://reviewboard.asterisk.org/r/4442/

........

Merged revisions 6483 from file:///srv/subversion/repos/testsuite/asterisk/trunk

chan_sip: Asterisk fails to re-activate an inactive media session when an

offer does not contain a=sendrecv

This test is to ensure that Asterisk correctly applies the direction of the

media stream when a=<sendonly|recvonly|inactive|sendrecv> is missing from the

offer's SDP. The expected behavior is for Asterisk to apply "sendrecv" as the

direction of the media stream when no direction attribute is present in an

offer's SDP.

According to RFC 4566 (Section 6. SDP Attributes): "If none of the attributes

"sendonly", "recvonly", "inactive", and "sendrecv" is present, "sendrecv"

SHOULD be assumed as the default for sessions that are not of the conference

type "broadcast" or "H332" [...]"

The test scenario:

1. From Phone A, send an offer to Phone B to establish a call

2. From Phone B, send an offer to Phone A to put the call on hold.

3. Observe that the MOH start event occurs.

4. From Phone B, send an offer to Phone A to 'un-hold' the call (ensure that

the direction attribute from the offer's SDP is omitted)

5. Observe that the MOH stop event occurs.

ASTERISK-24824 #close

Reported By: Ashley Sanders

Review: https://reviewboard.asterisk.org/r/4442/

res/res_pjsip_sdp_rtp: Revert portion of r432195

Unfortunately, while initial testing with ConfBridge did not reproduce the

audio problem alluded to in the comment in res_pjsip_sdp_rtp, further testing

did show that bridge_softmix and/or ConfBridge has a severe problem bridging

two or more participants at different sampling rates. Sometimes, it even picks

odd sampling rates that cause hideous audio problems.

This patch backs out the offending portion of the code until the issues in

the affected bridging modules can be more properly analyzed.

ASTERISK-24841

........

Merged revisions 432423 from http://svn.asterisk.org/svn/asterisk/branches/13

........

Merged revisions 432425 from file:///srv/subversion/repos/asterisk/trunk

res/res_pjsip_sdp_rtp: Revert portion of r432195

Unfortunately, while initial testing with ConfBridge did not reproduce the

audio problem alluded to in the comment in res_pjsip_sdp_rtp, further testing

did show that bridge_softmix and/or ConfBridge has a severe problem bridging

two or more participants at different sampling rates. Sometimes, it even picks

odd sampling rates that cause hideous audio problems.

This patch backs out the offending portion of the code until the issues in

the affected bridging modules can be more properly analyzed.

ASTERISK-24841

........

Merged revisions 432423 from http://svn.asterisk.org/svn/asterisk/branches/13

........

Merged revisions 432425 from file:///srv/subversion/repos/asterisk/trunk

res/res_pjsip_sdp_rtp: Revert portion of r432195

Unfortunately, while initial testing with ConfBridge did not reproduce the

audio problem alluded to in the comment in res_pjsip_sdp_rtp, further testing

did show that bridge_softmix and/or ConfBridge has a severe problem bridging

two or more participants at different sampling rates. Sometimes, it even picks

odd sampling rates that cause hideous audio problems.

This patch backs out the offending portion of the code until the issues in

the affected bridging modules can be more properly analyzed.

ASTERISK-24841

........

Merged revisions 432423 from http://svn.asterisk.org/svn/asterisk/branches/13

........

Merged revisions 432425 from file:///srv/subversion/repos/asterisk/trunk

res/res_pjsip_sdp_rtp: Revert portion of r432195

Unfortunately, while initial testing with ConfBridge did not reproduce the

audio problem alluded to in the comment in res_pjsip_sdp_rtp, further testing

did show that bridge_softmix and/or ConfBridge has a severe problem bridging

two or more participants at different sampling rates. Sometimes, it even picks

odd sampling rates that cause hideous audio problems.

This patch backs out the offending portion of the code until the issues in

the affected bridging modules can be more properly analyzed.

ASTERISK-24841

........

Merged revisions 432423 from http://svn.asterisk.org/svn/asterisk/branches/13

........

Merged revisions 432425 from file:///srv/subversion/repos/asterisk/trunk

res/res_pjsip_sdp_rtp: Revert portion of r432195

Unfortunately, while initial testing with ConfBridge did not reproduce the

audio problem alluded to in the comment in res_pjsip_sdp_rtp, further testing

did show that bridge_softmix and/or ConfBridge has a severe problem bridging

two or more participants at different sampling rates. Sometimes, it even picks

odd sampling rates that cause hideous audio problems.

This patch backs out the offending portion of the code until the issues in

the affected bridging modules can be more properly analyzed.

ASTERISK-24841

........

Merged revisions 432423 from http://svn.asterisk.org/svn/asterisk/branches/13

........

Merged revisions 432425 from file:///srv/subversion/repos/asterisk/trunk

res/res_pjsip_sdp_rtp: Revert portion of r432195

Unfortunately, while initial testing with ConfBridge did not reproduce the

audio problem alluded to in the comment in res_pjsip_sdp_rtp, further testing

did show that bridge_softmix and/or ConfBridge has a severe problem bridging

two or more participants at different sampling rates. Sometimes, it even picks

odd sampling rates that cause hideous audio problems.

This patch backs out the offending portion of the code until the issues in

the affected bridging modules can be more properly analyzed.

ASTERISK-24841

........

Merged revisions 432423 from http://svn.asterisk.org/svn/asterisk/branches/13

........

Merged revisions 432425 from file:///srv/subversion/repos/asterisk/trunk

res/res_pjsip_sdp_rtp: Revert portion of r432195

Unfortunately, while initial testing with ConfBridge did not reproduce the

audio problem alluded to in the comment in res_pjsip_sdp_rtp, further testing

did show that bridge_softmix and/or ConfBridge has a severe problem bridging

two or more participants at different sampling rates. Sometimes, it even picks

odd sampling rates that cause hideous audio problems.

This patch backs out the offending portion of the code until the issues in

the affected bridging modules can be more properly analyzed.

ASTERISK-24841

........

Merged revisions 432423 from http://svn.asterisk.org/svn/asterisk/branches/13

........

Merged revisions 432425 from file:///srv/subversion/repos/asterisk/trunk

res/res_pjsip_sdp_rtp: Revert portion of r432195

Unfortunately, while initial testing with ConfBridge did not reproduce the

audio problem alluded to in the comment in res_pjsip_sdp_rtp, further testing

did show that bridge_softmix and/or ConfBridge has a severe problem bridging

two or more participants at different sampling rates. Sometimes, it even picks

odd sampling rates that cause hideous audio problems.

This patch backs out the offending portion of the code until the issues in

the affected bridging modules can be more properly analyzed.

ASTERISK-24841

........

Merged revisions 432423 from http://svn.asterisk.org/svn/asterisk/branches/13

........

Merged revisions 432425 from file:///srv/subversion/repos/asterisk/trunk

res/res_pjsip_sdp_rtp: Revert portion of r432195

Unfortunately, while initial testing with ConfBridge did not reproduce the

audio problem alluded to in the comment in res_pjsip_sdp_rtp, further testing

did show that bridge_softmix and/or ConfBridge has a severe problem bridging

two or more participants at different sampling rates. Sometimes, it even picks

odd sampling rates that cause hideous audio problems.

This patch backs out the offending portion of the code until the issues in

the affected bridging modules can be more properly analyzed.

ASTERISK-24841

........

Merged revisions 432423 from file:///srv/subversion/repos/asterisk/branches/13

res/res_pjsip_sdp_rtp: Revert portion of r432195

Unfortunately, while initial testing with ConfBridge did not reproduce the

audio problem alluded to in the comment in res_pjsip_sdp_rtp, further testing

did show that bridge_softmix and/or ConfBridge has a severe problem bridging

two or more participants at different sampling rates. Sometimes, it even picks

odd sampling rates that cause hideous audio problems.

This patch backs out the offending portion of the code until the issues in

the affected bridging modules can be more properly analyzed.

ASTERISK-24841

........

Merged revisions 432423 from http://svn.asterisk.org/svn/asterisk/branches/13

........

Merged revisions 432425 from file:///srv/subversion/repos/asterisk/trunk

res/res_pjsip_sdp_rtp: Revert portion of r432195

Unfortunately, while initial testing with ConfBridge did not reproduce the

audio problem alluded to in the comment in res_pjsip_sdp_rtp, further testing

did show that bridge_softmix and/or ConfBridge has a severe problem bridging

two or more participants at different sampling rates. Sometimes, it even picks

odd sampling rates that cause hideous audio problems.

This patch backs out the offending portion of the code until the issues in

the affected bridging modules can be more properly analyzed.

ASTERISK-24841

........

Merged revisions 432423 from http://svn.asterisk.org/svn/asterisk/branches/13

........

Merged revisions 432425 from file:///srv/subversion/repos/asterisk/trunk

res/res_pjsip_sdp_rtp: Revert portion of r432195

Unfortunately, while initial testing with ConfBridge did not reproduce the

audio problem alluded to in the comment in res_pjsip_sdp_rtp, further testing

did show that bridge_softmix and/or ConfBridge has a severe problem bridging

two or more participants at different sampling rates. Sometimes, it even picks

odd sampling rates that cause hideous audio problems.

This patch backs out the offending portion of the code until the issues in

the affected bridging modules can be more properly analyzed.

ASTERISK-24841

........

Merged revisions 432423 from http://svn.asterisk.org/svn/asterisk/branches/13

........

Merged revisions 432425 from file:///srv/subversion/repos/asterisk/trunk

res/res_pjsip_sdp_rtp: Revert portion of r432195

Unfortunately, while initial testing with ConfBridge did not reproduce the

audio problem alluded to in the comment in res_pjsip_sdp_rtp, further testing

did show that bridge_softmix and/or ConfBridge has a severe problem bridging

two or more participants at different sampling rates. Sometimes, it even picks

odd sampling rates that cause hideous audio problems.

This patch backs out the offending portion of the code until the issues in

the affected bridging modules can be more properly analyzed.

ASTERISK-24841

........

Merged revisions 432423 from http://svn.asterisk.org/svn/asterisk/branches/13

........

Merged revisions 432425 from file:///srv/subversion/repos/asterisk/trunk

res/res_pjsip_sdp_rtp: Revert portion of r432195

Unfortunately, while initial testing with ConfBridge did not reproduce the

audio problem alluded to in the comment in res_pjsip_sdp_rtp, further testing

did show that bridge_softmix and/or ConfBridge has a severe problem bridging

two or more participants at different sampling rates. Sometimes, it even picks

odd sampling rates that cause hideous audio problems.

This patch backs out the offending portion of the code until the issues in

the affected bridging modules can be more properly analyzed.

ASTERISK-24841

........

Merged revisions 432423 from http://svn.asterisk.org/svn/asterisk/branches/13

........

Merged revisions 432425 from file:///srv/subversion/repos/asterisk/trunk

res/res_pjsip_sdp_rtp: Revert portion of r432195

Unfortunately, while initial testing with ConfBridge did not reproduce the

audio problem alluded to in the comment in res_pjsip_sdp_rtp, further testing

did show that bridge_softmix and/or ConfBridge has a severe problem bridging

two or more participants at different sampling rates. Sometimes, it even picks

odd sampling rates that cause hideous audio problems.

This patch backs out the offending portion of the code until the issues in

the affected bridging modules can be more properly analyzed.

ASTERISK-24841

........

Merged revisions 432423 from http://svn.asterisk.org/svn/asterisk/branches/13

res/res_pjsip_sdp_rtp: Revert portion of r432195

Unfortunately, while initial testing with ConfBridge did not reproduce the

audio problem alluded to in the comment in res_pjsip_sdp_rtp, further testing

did show that bridge_softmix and/or ConfBridge has a severe problem bridging

two or more participants at different sampling rates. Sometimes, it even picks

odd sampling rates that cause hideous audio problems.

This patch backs out the offending portion of the code until the issues in

the affected bridging modules can be more properly analyzed.

ASTERISK-24841

........

Merged revisions 432423 from http://svn.asterisk.org/svn/asterisk/branches/13

res/res_pjsip_sdp_rtp: Revert portion of r432195

Unfortunately, while initial testing with ConfBridge did not reproduce the

audio problem alluded to in the comment in res_pjsip_sdp_rtp, further testing

did show that bridge_softmix and/or ConfBridge has a severe problem bridging

two or more participants at different sampling rates. Sometimes, it even picks

odd sampling rates that cause hideous audio problems.

This patch backs out the offending portion of the code until the issues in

the affected bridging modules can be more properly analyzed.

ASTERISK-24841

    • -3
    • +22
    /branches/13/res/res_pjsip_sdp_rtp.c
ARI: Fix crash if integer values used in JSON payload 'variables' object.

Sending the following ARI commands caused Asterisk to crash if the JSON

body 'variables' object passes values of types other than strings.

POST /ari/channels

POST /ari/channels/{channelid}

PUT /ari/endpoints/sendMessage

PUT /ari/endpoints/{tech}/{resource}/sendMessage

* Eliminated RAII_VAR usage in ast_ari_channels_originate_with_id(),

ast_ari_channels_originate(), ast_ari_endpoints_send_message(), and

ast_ari_endpoints_send_message_to_endpoint().

ASTERISK-24751 #close

Reported by: jeffrey putnam

Review: https://reviewboard.asterisk.org/r/4447/

........

Merged revisions 432404 from http://svn.asterisk.org/svn/asterisk/branches/13

........

Merged revisions 432405 from file:///srv/subversion/repos/asterisk/trunk

    • -7
    • +22
    /team/rmudgett/bridge_tasks/main/json.c
ARI: Fix crash if integer values used in JSON payload 'variables' object.

Sending the following ARI commands caused Asterisk to crash if the JSON

body 'variables' object passes values of types other than strings.

POST /ari/channels

POST /ari/channels/{channelid}

PUT /ari/endpoints/sendMessage

PUT /ari/endpoints/{tech}/{resource}/sendMessage

* Eliminated RAII_VAR usage in ast_ari_channels_originate_with_id(),

ast_ari_channels_originate(), ast_ari_endpoints_send_message(), and

ast_ari_endpoints_send_message_to_endpoint().

ASTERISK-24751 #close

Reported by: jeffrey putnam

Review: https://reviewboard.asterisk.org/r/4447/

........

Merged revisions 432404 from http://svn.asterisk.org/svn/asterisk/branches/13

........

Merged revisions 432405 from file:///srv/subversion/repos/asterisk/trunk

ARI: Fix crash if integer values used in JSON payload 'variables' object.

Sending the following ARI commands caused Asterisk to crash if the JSON

body 'variables' object passes values of types other than strings.

POST /ari/channels

POST /ari/channels/{channelid}

PUT /ari/endpoints/sendMessage

PUT /ari/endpoints/{tech}/{resource}/sendMessage

* Eliminated RAII_VAR usage in ast_ari_channels_originate_with_id(),

ast_ari_channels_originate(), ast_ari_endpoints_send_message(), and

ast_ari_endpoints_send_message_to_endpoint().

ASTERISK-24751 #close

Reported by: jeffrey putnam

Review: https://reviewboard.asterisk.org/r/4447/

........

Merged revisions 432404 from http://svn.asterisk.org/svn/asterisk/branches/13

........

Merged revisions 432405 from file:///srv/subversion/repos/asterisk/trunk

    • -7
    • +22
    /team/mmichelson/queue_bugbug/main/json.c
ARI: Fix crash if integer values used in JSON payload 'variables' object.

Sending the following ARI commands caused Asterisk to crash if the JSON

body 'variables' object passes values of types other than strings.

POST /ari/channels

POST /ari/channels/{channelid}

PUT /ari/endpoints/sendMessage

PUT /ari/endpoints/{tech}/{resource}/sendMessage

* Eliminated RAII_VAR usage in ast_ari_channels_originate_with_id(),

ast_ari_channels_originate(), ast_ari_endpoints_send_message(), and

ast_ari_endpoints_send_message_to_endpoint().

ASTERISK-24751 #close

Reported by: jeffrey putnam

Review: https://reviewboard.asterisk.org/r/4447/

........

Merged revisions 432404 from http://svn.asterisk.org/svn/asterisk/branches/13

........

Merged revisions 432405 from file:///srv/subversion/repos/asterisk/trunk

ARI: Fix crash if integer values used in JSON payload 'variables' object.

Sending the following ARI commands caused Asterisk to crash if the JSON

body 'variables' object passes values of types other than strings.

POST /ari/channels

POST /ari/channels/{channelid}

PUT /ari/endpoints/sendMessage

PUT /ari/endpoints/{tech}/{resource}/sendMessage

* Eliminated RAII_VAR usage in ast_ari_channels_originate_with_id(),

ast_ari_channels_originate(), ast_ari_endpoints_send_message(), and

ast_ari_endpoints_send_message_to_endpoint().

ASTERISK-24751 #close

Reported by: jeffrey putnam

Review: https://reviewboard.asterisk.org/r/4447/

........

Merged revisions 432404 from http://svn.asterisk.org/svn/asterisk/branches/13

........

Merged revisions 432405 from file:///srv/subversion/repos/asterisk/trunk

ARI: Fix crash if integer values used in JSON payload 'variables' object.

Sending the following ARI commands caused Asterisk to crash if the JSON

body 'variables' object passes values of types other than strings.

POST /ari/channels

POST /ari/channels/{channelid}

PUT /ari/endpoints/sendMessage

PUT /ari/endpoints/{tech}/{resource}/sendMessage

* Eliminated RAII_VAR usage in ast_ari_channels_originate_with_id(),

ast_ari_channels_originate(), ast_ari_endpoints_send_message(), and

ast_ari_endpoints_send_message_to_endpoint().

ASTERISK-24751 #close

Reported by: jeffrey putnam

Review: https://reviewboard.asterisk.org/r/4447/

........

Merged revisions 432404 from http://svn.asterisk.org/svn/asterisk/branches/13

........

Merged revisions 432405 from file:///srv/subversion/repos/asterisk/trunk

ARI: Fix crash if integer values used in JSON payload 'variables' object.

Sending the following ARI commands caused Asterisk to crash if the JSON

body 'variables' object passes values of types other than strings.

POST /ari/channels

POST /ari/channels/{channelid}

PUT /ari/endpoints/sendMessage

PUT /ari/endpoints/{tech}/{resource}/sendMessage

* Eliminated RAII_VAR usage in ast_ari_channels_originate_with_id(),

ast_ari_channels_originate(), ast_ari_endpoints_send_message(), and

ast_ari_endpoints_send_message_to_endpoint().

ASTERISK-24751 #close

Reported by: jeffrey putnam

Review: https://reviewboard.asterisk.org/r/4447/

........

Merged revisions 432404 from http://svn.asterisk.org/svn/asterisk/branches/13

........

Merged revisions 432405 from file:///srv/subversion/repos/asterisk/trunk

ARI: Fix crash if integer values used in JSON payload 'variables' object.

Sending the following ARI commands caused Asterisk to crash if the JSON

body 'variables' object passes values of types other than strings.

POST /ari/channels

POST /ari/channels/{channelid}

PUT /ari/endpoints/sendMessage

PUT /ari/endpoints/{tech}/{resource}/sendMessage

* Eliminated RAII_VAR usage in ast_ari_channels_originate_with_id(),

ast_ari_channels_originate(), ast_ari_endpoints_send_message(), and

ast_ari_endpoints_send_message_to_endpoint().

ASTERISK-24751 #close

Reported by: jeffrey putnam

Review: https://reviewboard.asterisk.org/r/4447/

........

Merged revisions 432404 from http://svn.asterisk.org/svn/asterisk/branches/13

........

Merged revisions 432405 from file:///srv/subversion/repos/asterisk/trunk