<p>Salah Ahmed has uploaded this change for <strong>review</strong>.</p><p><a href="https://gerrit.asterisk.org/c/asterisk/+/13501">View Change</a></p><pre style="font-family: monospace,monospace; white-space: pre-wrap;">asymmetric_rtp_codec dialplan function<br><br>Implement a dialplan function for asymmetric_rtp_codec. We<br>can change this flag on per call basis.<br><br>ASTERISK-28668<br><br>Change-Id: I600d5ac1c6c2b47a7a0da6425aea559f1aa61c7c<br>---<br>M channels/chan_pjsip.c<br>M channels/pjsip/dialplan_functions.c<br>M channels/pjsip/include/dialplan_functions.h<br>M include/asterisk/res_pjsip_session.h<br>M res/res_pjsip_sdp_rtp.c<br>M res/res_pjsip_session.c<br>6 files changed, 113 insertions(+), 4 deletions(-)<br><br></pre><pre style="font-family: monospace,monospace; white-space: pre-wrap;">git pull ssh://gerrit.asterisk.org:29418/asterisk refs/changes/01/13501/1</pre><pre style="font-family: monospace,monospace; white-space: pre-wrap;"><span>diff --git a/channels/chan_pjsip.c b/channels/chan_pjsip.c</span><br><span>index 33abe03..422470c 100644</span><br><span>--- a/channels/chan_pjsip.c</span><br><span>+++ b/channels/chan_pjsip.c</span><br><span>@@ -770,7 +770,7 @@</span><br><span> * Therefore we need to update the native format to the current</span><br><span> * raw read format BEFORE the native format check</span><br><span> */</span><br><span style="color: hsl(0, 100%, 40%);">- if (!session->endpoint->asymmetric_rtp_codec &&</span><br><span style="color: hsl(120, 100%, 40%);">+ if (!session->asymmetric_rtp_codec &&</span><br><span> ast_format_cmp(ast_channel_rawwriteformat(ast), f->subclass.format) == AST_FORMAT_CMP_NOT_EQUAL) {</span><br><span> struct ast_format_cap *caps;</span><br><span> </span><br><span>@@ -2875,6 +2875,12 @@</span><br><span> .write = pjsip_acf_dtmf_mode_write</span><br><span> };</span><br><span> </span><br><span style="color: hsl(120, 100%, 40%);">+static struct ast_custom_function asymmetric_rtp_codec_function = {</span><br><span style="color: hsl(120, 100%, 40%);">+ .name = "PJSIP_ASYMMETRIC_RTP_CODEC",</span><br><span style="color: hsl(120, 100%, 40%);">+ .read = pjsip_acf_asymmetric_rtp_codec_read,</span><br><span style="color: hsl(120, 100%, 40%);">+ .write = pjsip_acf_asymmetric_rtp_codec_write</span><br><span style="color: hsl(120, 100%, 40%);">+};</span><br><span style="color: hsl(120, 100%, 40%);">+</span><br><span> static struct ast_custom_function moh_passthrough_function = {</span><br><span> .name = "PJSIP_MOH_PASSTHROUGH",</span><br><span> .read = pjsip_acf_moh_passthrough_read,</span><br><span>@@ -2935,6 +2941,11 @@</span><br><span> goto end;</span><br><span> }</span><br><span> </span><br><span style="color: hsl(120, 100%, 40%);">+ if (ast_custom_function_register(&asymmetric_rtp_codec_function)) {</span><br><span style="color: hsl(120, 100%, 40%);">+ ast_log(LOG_WARNING, "Unable to register PJSIP_ASYMMETRIC_RTP_CODEC dialplan function\n");</span><br><span style="color: hsl(120, 100%, 40%);">+ goto end;</span><br><span style="color: hsl(120, 100%, 40%);">+ }</span><br><span style="color: hsl(120, 100%, 40%);">+</span><br><span> if (ast_custom_function_register(&moh_passthrough_function)) {</span><br><span> ast_log(LOG_WARNING, "Unable to register PJSIP_MOH_PASSTHROUGH dialplan function\n");</span><br><span> goto end;</span><br><span>@@ -3000,6 +3011,7 @@</span><br><span> ast_sip_session_unregister_supplement(&chan_pjsip_supplement);</span><br><span> ast_sip_session_unregister_supplement(&call_pickup_supplement);</span><br><span> ast_custom_function_unregister(&dtmf_mode_function);</span><br><span style="color: hsl(120, 100%, 40%);">+ ast_custom_function_unregister(&asymmetric_rtp_codec_function);</span><br><span> ast_custom_function_unregister(&moh_passthrough_function);</span><br><span> ast_custom_function_unregister(&media_offer_function);</span><br><span> ast_custom_function_unregister(&chan_pjsip_dial_contacts_function);</span><br><span>@@ -3026,6 +3038,7 @@</span><br><span> ast_sip_session_unregister_supplement(&call_pickup_supplement);</span><br><span> </span><br><span> ast_custom_function_unregister(&dtmf_mode_function);</span><br><span style="color: hsl(120, 100%, 40%);">+ ast_custom_function_unregister(&asymmetric_rtp_codec_function);</span><br><span> ast_custom_function_unregister(&moh_passthrough_function);</span><br><span> ast_custom_function_unregister(&media_offer_function);</span><br><span> ast_custom_function_unregister(&chan_pjsip_dial_contacts_function);</span><br><span>diff --git a/channels/pjsip/dialplan_functions.c b/channels/pjsip/dialplan_functions.c</span><br><span>index 3a6f0a4..74e7ace 100644</span><br><span>--- a/channels/pjsip/dialplan_functions.c</span><br><span>+++ b/channels/pjsip/dialplan_functions.c</span><br><span>@@ -80,6 +80,18 @@</span><br><span> <para>This function uses the same DTMF mode naming as the dtmf_mode configuration option</para></span><br><span> </description></span><br><span> </function></span><br><span style="color: hsl(120, 100%, 40%);">+<function name="PJSIP_ASYMMETRIC_RTP_CODEC" language="en_US"></span><br><span style="color: hsl(120, 100%, 40%);">+ <synopsis></span><br><span style="color: hsl(120, 100%, 40%);">+ Get or change the asymmethric rtp codec.</span><br><span style="color: hsl(120, 100%, 40%);">+ </synopsis></span><br><span style="color: hsl(120, 100%, 40%);">+ <syntax></span><br><span style="color: hsl(120, 100%, 40%);">+ </syntax></span><br><span style="color: hsl(120, 100%, 40%);">+ <description></span><br><span style="color: hsl(120, 100%, 40%);">+ <para>When read, returns the current asymmetric_rtp_codec mode</para></span><br><span style="color: hsl(120, 100%, 40%);">+ <para>When written, sets the current asymmetric_rtp_codec mode</para></span><br><span style="color: hsl(120, 100%, 40%);">+ <para>Allow the sending and receiving RTP codec to differ</para></span><br><span style="color: hsl(120, 100%, 40%);">+ </description></span><br><span style="color: hsl(120, 100%, 40%);">+</function></span><br><span> <function name="PJSIP_MOH_PASSTHROUGH" language="en_US"></span><br><span> <synopsis></span><br><span> Get or change the on-hold behavior for a SIP call.</span><br><span>@@ -1316,6 +1328,62 @@</span><br><span> return 0;</span><br><span> }</span><br><span> </span><br><span style="color: hsl(120, 100%, 40%);">+int pjsip_acf_asymmetric_rtp_codec_read(struct ast_channel *chan, const char *cmd, char *data, char *buf, size_t len)</span><br><span style="color: hsl(120, 100%, 40%);">+{</span><br><span style="color: hsl(120, 100%, 40%);">+ struct ast_sip_channel_pvt *channel;</span><br><span style="color: hsl(120, 100%, 40%);">+</span><br><span style="color: hsl(120, 100%, 40%);">+ if (!chan) {</span><br><span style="color: hsl(120, 100%, 40%);">+ ast_log(LOG_WARNING, "No channel was provided to %s function.\n", cmd);</span><br><span style="color: hsl(120, 100%, 40%);">+ return -1;</span><br><span style="color: hsl(120, 100%, 40%);">+ }</span><br><span style="color: hsl(120, 100%, 40%);">+</span><br><span style="color: hsl(120, 100%, 40%);">+ ast_channel_lock(chan);</span><br><span style="color: hsl(120, 100%, 40%);">+ if (strcmp(ast_channel_tech(chan)->type, "PJSIP")) {</span><br><span style="color: hsl(120, 100%, 40%);">+ ast_log(LOG_WARNING, "Cannot call %s on a non-PJSIP channel\n", cmd);</span><br><span style="color: hsl(120, 100%, 40%);">+ ast_channel_unlock(chan);</span><br><span style="color: hsl(120, 100%, 40%);">+ return -1;</span><br><span style="color: hsl(120, 100%, 40%);">+ }</span><br><span style="color: hsl(120, 100%, 40%);">+</span><br><span style="color: hsl(120, 100%, 40%);">+ channel = ast_channel_tech_pvt(chan);</span><br><span style="color: hsl(120, 100%, 40%);">+</span><br><span style="color: hsl(120, 100%, 40%);">+ if (len < 3) {</span><br><span style="color: hsl(120, 100%, 40%);">+ ast_log(LOG_WARNING, "%s: buffer too small\n", cmd);</span><br><span style="color: hsl(120, 100%, 40%);">+ ast_channel_unlock(chan);</span><br><span style="color: hsl(120, 100%, 40%);">+ return -1;</span><br><span style="color: hsl(120, 100%, 40%);">+ }</span><br><span style="color: hsl(120, 100%, 40%);">+</span><br><span style="color: hsl(120, 100%, 40%);">+ channel = ast_channel_tech_pvt(chan);</span><br><span style="color: hsl(120, 100%, 40%);">+ strncpy(buf, AST_YESNO(channel->session->asymmetric_rtp_codec), len);</span><br><span style="color: hsl(120, 100%, 40%);">+</span><br><span style="color: hsl(120, 100%, 40%);">+ ast_channel_unlock(chan);</span><br><span style="color: hsl(120, 100%, 40%);">+ return 0;</span><br><span style="color: hsl(120, 100%, 40%);">+}</span><br><span style="color: hsl(120, 100%, 40%);">+</span><br><span style="color: hsl(120, 100%, 40%);">+int pjsip_acf_asymmetric_rtp_codec_write(struct ast_channel *chan, const char *cmd, char *data, const char *value)</span><br><span style="color: hsl(120, 100%, 40%);">+{</span><br><span style="color: hsl(120, 100%, 40%);">+ struct ast_sip_channel_pvt *channel;</span><br><span style="color: hsl(120, 100%, 40%);">+</span><br><span style="color: hsl(120, 100%, 40%);">+ if (!chan) {</span><br><span style="color: hsl(120, 100%, 40%);">+ ast_log(LOG_WARNING, "No channel was provided to %s function.\n", cmd);</span><br><span style="color: hsl(120, 100%, 40%);">+ return -1;</span><br><span style="color: hsl(120, 100%, 40%);">+ }</span><br><span style="color: hsl(120, 100%, 40%);">+</span><br><span style="color: hsl(120, 100%, 40%);">+ ast_channel_lock(chan);</span><br><span style="color: hsl(120, 100%, 40%);">+ if (strcmp(ast_channel_tech(chan)->type, "PJSIP")) {</span><br><span style="color: hsl(120, 100%, 40%);">+ ast_log(LOG_WARNING, "Cannot call %s on a non-PJSIP channel\n", cmd);</span><br><span style="color: hsl(120, 100%, 40%);">+ ast_channel_unlock(chan);</span><br><span style="color: hsl(120, 100%, 40%);">+ return -1;</span><br><span style="color: hsl(120, 100%, 40%);">+ }</span><br><span style="color: hsl(120, 100%, 40%);">+</span><br><span style="color: hsl(120, 100%, 40%);">+ channel = ast_channel_tech_pvt(chan);</span><br><span style="color: hsl(120, 100%, 40%);">+ channel->session->asymmetric_rtp_codec = ast_true(value);</span><br><span style="color: hsl(120, 100%, 40%);">+</span><br><span style="color: hsl(120, 100%, 40%);">+ ast_channel_unlock(chan);</span><br><span style="color: hsl(120, 100%, 40%);">+</span><br><span style="color: hsl(120, 100%, 40%);">+ return 0;</span><br><span style="color: hsl(120, 100%, 40%);">+}</span><br><span style="color: hsl(120, 100%, 40%);">+</span><br><span style="color: hsl(120, 100%, 40%);">+</span><br><span> int pjsip_acf_moh_passthrough_read(struct ast_channel *chan, const char *cmd, char *data, char *buf, size_t len)</span><br><span> {</span><br><span> struct ast_sip_channel_pvt *channel;</span><br><span>diff --git a/channels/pjsip/include/dialplan_functions.h b/channels/pjsip/include/dialplan_functions.h</span><br><span>index d0bf130..dd81055 100644</span><br><span>--- a/channels/pjsip/include/dialplan_functions.h</span><br><span>+++ b/channels/pjsip/include/dialplan_functions.h</span><br><span>@@ -73,6 +73,31 @@</span><br><span> int pjsip_acf_dtmf_mode_write(struct ast_channel *chan, const char *cmd, char *data, const char *value);</span><br><span> </span><br><span> /*!</span><br><span style="color: hsl(120, 100%, 40%);">+ * \brief PJSIP_ASYMMETRIC_RTP_CODEC function read callback</span><br><span style="color: hsl(120, 100%, 40%);">+ * \param chan The channel the function is called on</span><br><span style="color: hsl(120, 100%, 40%);">+ * \param cmd The name of the function</span><br><span style="color: hsl(120, 100%, 40%);">+ * \param data Arguments passed to the function</span><br><span style="color: hsl(120, 100%, 40%);">+ * \param buf Out buffer that should be populated with the data</span><br><span style="color: hsl(120, 100%, 40%);">+ * \param len Size of the buffer</span><br><span style="color: hsl(120, 100%, 40%);">+ *</span><br><span style="color: hsl(120, 100%, 40%);">+ * \retval 0 on success</span><br><span style="color: hsl(120, 100%, 40%);">+ * \retval -1 on failure</span><br><span style="color: hsl(120, 100%, 40%);">+ */</span><br><span style="color: hsl(120, 100%, 40%);">+int pjsip_acf_asymmetric_rtp_codec_read(struct ast_channel *chan, const char *cmd, char *data, char *buf, size_t len);</span><br><span style="color: hsl(120, 100%, 40%);">+</span><br><span style="color: hsl(120, 100%, 40%);">+/*!</span><br><span style="color: hsl(120, 100%, 40%);">+ * \brief PJSIP_ASYMMETRIC_RTP_CODEC function write callback</span><br><span style="color: hsl(120, 100%, 40%);">+ * \param chan The channel the function is called on</span><br><span style="color: hsl(120, 100%, 40%);">+ * \param cmd The name of the function</span><br><span style="color: hsl(120, 100%, 40%);">+ * \param data Arguments passed to the function</span><br><span style="color: hsl(120, 100%, 40%);">+ * \param value Value to be set by the function</span><br><span style="color: hsl(120, 100%, 40%);">+ *</span><br><span style="color: hsl(120, 100%, 40%);">+ * \retval 0 on success</span><br><span style="color: hsl(120, 100%, 40%);">+ * \retval -1 on failure</span><br><span style="color: hsl(120, 100%, 40%);">+ */</span><br><span style="color: hsl(120, 100%, 40%);">+int pjsip_acf_asymmetric_rtp_codec_write(struct ast_channel *chan, const char *cmd, char *data, const char *value);</span><br><span style="color: hsl(120, 100%, 40%);">+</span><br><span style="color: hsl(120, 100%, 40%);">+/*!</span><br><span> * \brief PJSIP_MOH_PASSTHROUGH function read callback</span><br><span> * \param chan The channel the function is called on</span><br><span> * \param cmd The name of the function</span><br><span>@@ -148,4 +173,4 @@</span><br><span> */</span><br><span> int pjsip_acf_parse_uri_read(struct ast_channel *chan, const char *cmd, char *data, char *buf, size_t len);</span><br><span> </span><br><span style="color: hsl(0, 100%, 40%);">-#endif /* _PJSIP_DIALPLAN_FUNCTIONS */</span><br><span>\ No newline at end of file</span><br><span style="color: hsl(120, 100%, 40%);">+#endif /* _PJSIP_DIALPLAN_FUNCTIONS */</span><br><span>diff --git a/include/asterisk/res_pjsip_session.h b/include/asterisk/res_pjsip_session.h</span><br><span>index 9ae1883..ba017a6 100644</span><br><span>--- a/include/asterisk/res_pjsip_session.h</span><br><span>+++ b/include/asterisk/res_pjsip_session.h</span><br><span>@@ -161,6 +161,8 @@</span><br><span> unsigned int defer_end:1;</span><br><span> /*! Session end (remote hangup) requested while termination deferred */</span><br><span> unsigned int ended_while_deferred:1;</span><br><span style="color: hsl(120, 100%, 40%);">+ /*! asymmetric rtp codec */</span><br><span style="color: hsl(120, 100%, 40%);">+ unsigned int asymmetric_rtp_codec:1;</span><br><span> /*! Whether to pass through hold and unhold using re-invites with recvonly and sendrecv */</span><br><span> unsigned int moh_passthrough:1;</span><br><span> /*! DTMF mode to use with this session, from endpoint but can change */</span><br><span>diff --git a/res/res_pjsip_sdp_rtp.c b/res/res_pjsip_sdp_rtp.c</span><br><span>index 98e9819..3baab40 100644</span><br><span>--- a/res/res_pjsip_sdp_rtp.c</span><br><span>+++ b/res/res_pjsip_sdp_rtp.c</span><br><span>@@ -241,7 +241,7 @@</span><br><span> }</span><br><span> </span><br><span> ast_rtp_instance_set_prop(session_media->rtp, AST_RTP_PROPERTY_NAT, session->endpoint->media.rtp.symmetric);</span><br><span style="color: hsl(0, 100%, 40%);">- ast_rtp_instance_set_prop(session_media->rtp, AST_RTP_PROPERTY_ASYMMETRIC_CODEC, session->endpoint->asymmetric_rtp_codec);</span><br><span style="color: hsl(120, 100%, 40%);">+ ast_rtp_instance_set_prop(session_media->rtp, AST_RTP_PROPERTY_ASYMMETRIC_CODEC, session->asymmetric_rtp_codec);</span><br><span> </span><br><span> if (!session->endpoint->media.rtp.ice_support && (ice = ast_rtp_instance_get_ice(session_media->rtp))) {</span><br><span> ice->stop(session_media->rtp);</span><br><span>@@ -419,7 +419,7 @@</span><br><span> * type and use only that. This ensures the core won't start sending</span><br><span> * out a format that we aren't currently sending.</span><br><span> */</span><br><span style="color: hsl(0, 100%, 40%);">- if (!session->endpoint->asymmetric_rtp_codec) {</span><br><span style="color: hsl(120, 100%, 40%);">+ if (!session->asymmetric_rtp_codec) {</span><br><span> struct ast_format *best;</span><br><span> </span><br><span> best = ast_format_cap_get_best_by_type(joint, media_type);</span><br><span>diff --git a/res/res_pjsip_session.c b/res/res_pjsip_session.c</span><br><span>index bbdb8d1..8e629e5 100644</span><br><span>--- a/res/res_pjsip_session.c</span><br><span>+++ b/res/res_pjsip_session.c</span><br><span>@@ -1494,6 +1494,7 @@</span><br><span> </span><br><span> session->dtmf = endpoint->dtmf;</span><br><span> session->moh_passthrough = endpoint->moh_passthrough;</span><br><span style="color: hsl(120, 100%, 40%);">+ session->asymmetric_rtp_codec = endpoint->asymmetric_rtp_codec;</span><br><span> </span><br><span> if (ast_sip_session_add_supplements(session)) {</span><br><span> /* Release the ref held by session->inv_session */</span><br><span></span><br></pre><p>To view, visit <a href="https://gerrit.asterisk.org/c/asterisk/+/13501">change 13501</a>. To unsubscribe, or for help writing mail filters, visit <a href="https://gerrit.asterisk.org/settings">settings</a>.</p><div itemscope itemtype="http://schema.org/EmailMessage"><div itemscope itemprop="action" itemtype="http://schema.org/ViewAction"><link itemprop="url" href="https://gerrit.asterisk.org/c/asterisk/+/13501"/><meta itemprop="name" content="View Change"/></div></div>
<div style="display:none"> Gerrit-Project: asterisk </div>
<div style="display:none"> Gerrit-Branch: 13 </div>
<div style="display:none"> Gerrit-Change-Id: I600d5ac1c6c2b47a7a0da6425aea559f1aa61c7c </div>
<div style="display:none"> Gerrit-Change-Number: 13501 </div>
<div style="display:none"> Gerrit-PatchSet: 1 </div>
<div style="display:none"> Gerrit-Owner: Salah Ahmed <txrubel@gmail.com> </div>
<div style="display:none"> Gerrit-MessageType: newchange </div>