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https://gitea.osmocom.org/cellular-infrastructure/osmo-mgw.git
synced 2025-10-23 08:12:01 +00:00
mgcp_trunk: remove audio_name and audio_payload
get rid of deprecated trunk parameters which seem to be leftovers from the old osmo-bsc_mgcp implementation. This is in particular audio_name and audio_payload in struct mgcp_trunk_config which allowed the user to "hardcode" an andio name and payload type via VTY configuration The removal of the struct members above also require a change to mgcp_codec.c. The code that is is never actively used and even causes wrong behavior when activated (set the no-transcoding flag in VTY). Since the code is removed also the unit tests also require to be changed to match the new behavior. Change-Id: Ia050ec3cd34b410dfe089c41b977ae3d5aed7354 Related: OS#2659
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@@ -9,8 +9,6 @@ struct mgcp_trunk {
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int trunk_type;
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char *audio_fmtp_extra;
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char *audio_name;
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int audio_payload;
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int audio_send_ptime;
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int audio_send_name;
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int audio_loop;
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@@ -280,36 +280,16 @@ error:
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* Helper function for mgcp_codec_decide() */
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static bool is_codec_compatible(const struct mgcp_endpoint *endp, const struct mgcp_rtp_codec *codec)
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{
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char codec_name[64];
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/* A codec name must be set, if not, this might mean that the codec
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* (payload type) that was assigned is unknown to us so we must stop
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* here. */
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if (!codec->subtype_name)
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return false;
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/* We now extract the codec_name (letters before the /, e.g. "GSM"
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* from the audio name that is stored in the trunk configuration.
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* We do not compare to the full audio_name because we expect that
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* "GSM", "GSM/8000" and "GSM/8000/1" are all compatible when the
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* audio name of the codec is set to "GSM" */
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if (sscanf(endp->trunk->audio_name, "%63[^/]/%*d/%*d", codec_name) < 1)
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return false;
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/* FIXME: implement meaningful checks to make sure that the given codec
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* is compatible with the given endpoint */
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/* Finally we check if the subtype_name we have generated from the
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* audio_name in the trunc struct patches the codec_name of the
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* given codec */
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if (strcasecmp(codec_name, codec->subtype_name) == 0)
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return true;
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/* FIXME: It is questinable that the method to pick a compatible
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* codec can work properly. Since this useses trunk->audio_name, as
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* a reference, which is set to "AMR/8000" permanently.
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* trunk->audio_name must be updated by the first connection that
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* has been made on an endpoint, so that the second connection
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* can make a meaningful decision here */
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return false;
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return true;
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}
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/*! Decide for one suitable codec
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@@ -114,12 +114,6 @@ static int config_write_mgcp(struct vty *vty)
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VTY_NEWLINE);
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} else
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vty_out(vty, " no rtp-patch%s", VTY_NEWLINE);
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if (trunk->audio_payload != -1)
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vty_out(vty, " sdp audio-payload number %d%s",
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trunk->audio_payload, VTY_NEWLINE);
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if (trunk->audio_name)
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vty_out(vty, " sdp audio-payload name %s%s",
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trunk->audio_name, VTY_NEWLINE);
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if (trunk->audio_fmtp_extra)
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vty_out(vty, " sdp audio fmtp-extra %s%s",
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trunk->audio_fmtp_extra, VTY_NEWLINE);
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@@ -611,13 +605,11 @@ DEFUN(cfg_mgcp_no_allow_transcoding,
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#define SDP_STR "SDP File related options\n"
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#define AUDIO_STR "Audio payload options\n"
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DEFUN(cfg_mgcp_sdp_payload_number,
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DEFUN_DEPRECATED(cfg_mgcp_sdp_payload_number,
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cfg_mgcp_sdp_payload_number_cmd,
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"sdp audio-payload number <0-255>",
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SDP_STR AUDIO_STR "Number\n" "Payload number\n")
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{
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unsigned int payload = atoi(argv[0]);
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g_cfg->virt_trunk->audio_payload = payload;
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return CMD_SUCCESS;
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}
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@@ -626,12 +618,11 @@ ALIAS_DEPRECATED(cfg_mgcp_sdp_payload_number,
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"sdp audio payload number <0-255>",
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SDP_STR AUDIO_STR AUDIO_STR "Number\n" "Payload number\n")
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DEFUN(cfg_mgcp_sdp_payload_name,
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DEFUN_DEPRECATED(cfg_mgcp_sdp_payload_name,
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cfg_mgcp_sdp_payload_name_cmd,
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"sdp audio-payload name NAME",
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SDP_STR AUDIO_STR "Name\n" "Payload name\n")
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{
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osmo_talloc_replace_string(g_cfg, &g_cfg->virt_trunk->audio_name, argv[0]);
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return CMD_SUCCESS;
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}
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@@ -856,10 +847,6 @@ static int config_write_trunk(struct vty *vty)
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llist_for_each_entry(trunk, &g_cfg->trunks, entry) {
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vty_out(vty, " trunk %d%s", trunk->trunk_nr, VTY_NEWLINE);
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vty_out(vty, " sdp audio-payload number %d%s",
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trunk->audio_payload, VTY_NEWLINE);
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vty_out(vty, " sdp audio-payload name %s%s",
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trunk->audio_name, VTY_NEWLINE);
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vty_out(vty, " %ssdp audio-payload send-ptime%s",
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trunk->audio_send_ptime ? "" : "no ", VTY_NEWLINE);
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vty_out(vty, " %ssdp audio-payload send-name%s",
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@@ -920,15 +907,11 @@ DEFUN(cfg_trunk_sdp_fmtp_extra,
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return CMD_SUCCESS;
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}
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DEFUN(cfg_trunk_payload_number,
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DEFUN_DEPRECATED(cfg_trunk_payload_number,
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cfg_trunk_payload_number_cmd,
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"sdp audio-payload number <0-255>",
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SDP_STR AUDIO_STR "Number\n" "Payload Number\n")
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{
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struct mgcp_trunk *trunk = vty->index;
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unsigned int payload = atoi(argv[0]);
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trunk->audio_payload = payload;
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return CMD_SUCCESS;
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}
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@@ -936,14 +919,11 @@ ALIAS_DEPRECATED(cfg_trunk_payload_number, cfg_trunk_payload_number_cmd_old,
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"sdp audio payload number <0-255>",
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SDP_STR AUDIO_STR AUDIO_STR "Number\n" "Payload Number\n")
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DEFUN(cfg_trunk_payload_name,
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DEFUN_DEPRECATED(cfg_trunk_payload_name,
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cfg_trunk_payload_name_cmd,
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"sdp audio-payload name NAME",
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SDP_STR AUDIO_STR "Payload\n" "Payload Name\n")
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{
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struct mgcp_trunk *trunk = vty->index;
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osmo_talloc_replace_string(g_cfg, &trunk->audio_name, argv[0]);
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return CMD_SUCCESS;
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}
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@@ -1458,11 +1458,9 @@ static void test_multilple_codec(void)
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OSMO_ASSERT(conn);
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OSMO_ASSERT(conn->end.codec->payload_type == 18);
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/* Allocate 5@mgw at select GSM.. */
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/* Allocate 5@mgw and let osmo-mgw pick a codec from the list */
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last_endpoint = -1;
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inp = create_msg(CRCX_MULT_GSM_EXACT, NULL);
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talloc_free(cfg->virt_trunk->audio_name);
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cfg->virt_trunk->audio_name = "GSM/8000";
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cfg->virt_trunk->no_audio_transcoding = 1;
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resp = mgcp_handle_message(cfg, inp);
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OSMO_ASSERT(get_conn_id_from_response(resp->data, conn_id,
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@@ -1474,7 +1472,7 @@ static void test_multilple_codec(void)
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endp = cfg->virt_trunk->endpoints[last_endpoint];
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conn = mgcp_conn_get_rtp(endp, conn_id);
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OSMO_ASSERT(conn);
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OSMO_ASSERT(conn->end.codec->payload_type == 3);
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OSMO_ASSERT(conn->end.codec->payload_type == 0);
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inp = create_msg(MDCX_NAT_DUMMY, conn_id);
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last_endpoint = -1;
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