sip: Implement MT call out to SIP
* Create a new handle * Send the invite * Have some state transitions * Allow to release a call in initial unconfirmed state, confirmed one with cancel and connected with bye * Add simple SDP parsing to find the rtpmap/codec that is used by gsm
This commit is contained in:
Holger Hans Peter Freyther
parent
2e36090fea
commit
2211c3ba56
16
src/call.h
16
src/call.h
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@ -10,6 +10,9 @@
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struct sip_agent;
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struct mncc_connection;
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struct nua_handle_s;
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struct call_leg;
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/**
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@ -62,10 +65,23 @@ struct call_leg {
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void (*release_call)(struct call_leg *);
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};
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enum sip_cc_state {
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SIP_CC_INITIAL,
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SIP_CC_DLG_CNFD,
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SIP_CC_CONNECTED,
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};
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struct sip_call_leg {
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/* base class */
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struct call_leg base;
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/* back pointer */
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struct sip_agent *agent;
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/* per instance members */
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struct nua_handle_s *nua_handle;
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enum sip_cc_state state;
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const char *wanted_codec;
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};
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enum mncc_cc_state {
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268
src/sip.c
268
src/sip.c
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@ -20,6 +20,12 @@
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#include "sip.h"
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#include "app.h"
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#include "call.h"
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#include "logging.h"
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#include <osmocom/core/utils.h>
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#include <sofia-sip/sdp.h>
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#include <talloc.h>
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@ -27,8 +33,270 @@
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extern void *tall_mncc_ctx;
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static bool extract_sdp(struct sip_call_leg *leg, const sip_t *sip)
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{
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sdp_connection_t *conn;
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sdp_session_t *sdp;
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sdp_parser_t *parser;
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sdp_media_t *media;
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const char *sdp_data;
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bool found_conn = false, found_map = false;
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if (!sip->sip_payload || !sip->sip_payload->pl_data) {
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LOGP(DSIP, LOGL_ERROR, "leg(%p) but no SDP file\n", leg);
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return false;
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}
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sdp_data = sip->sip_payload->pl_data;
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parser = sdp_parse(NULL, sdp_data, strlen(sdp_data), 0);
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if (!parser) {
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LOGP(DSIP, LOGL_ERROR, "leg(%p) failed to parse SDP\n",
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leg);
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return false;
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}
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sdp = sdp_session(parser);
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if (!sdp) {
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LOGP(DSIP, LOGL_ERROR, "leg(%p) no sdp session\n", leg);
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sdp_parser_free(parser);
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return false;
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}
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for (conn = sdp->sdp_connection; conn; conn = conn->c_next) {
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struct in_addr addr;
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if (conn->c_addrtype != sdp_addr_ip4)
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continue;
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inet_aton(conn->c_address, &addr);
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leg->base.ip = addr.s_addr;
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found_conn = true;
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break;
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}
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for (media = sdp->sdp_media; media; media = media->m_next) {
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sdp_rtpmap_t *map;
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if (media->m_proto != sdp_proto_rtp)
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continue;
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if (media->m_type != sdp_media_audio)
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continue;
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for (map = media->m_rtpmaps; map; map = map->rm_next) {
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if (strcasecmp(map->rm_encoding, leg->wanted_codec) != 0)
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continue;
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leg->base.port = media->m_port;
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leg->base.payload_type = map->rm_pt;
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found_map = true;
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break;
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}
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if (found_map)
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break;
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}
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if (!found_conn || !found_map) {
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LOGP(DSIP, LOGL_ERROR, "leg(%p) did not find %d/%d\n",
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leg, found_conn, found_map);
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sdp_parser_free(parser);
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return false;
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}
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sdp_parser_free(parser);
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return true;
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}
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static void call_progress(struct sip_call_leg *leg, const sip_t *sip)
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{
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struct call_leg *other = call_leg_other(&leg->base);
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if (!other)
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return;
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LOGP(DSIP, LOGL_NOTICE, "leg(%p) is now rining.\n", leg);
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other->ring_call(other);
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}
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static void call_connect(struct sip_call_leg *leg, const sip_t *sip)
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{
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/* extract SDP file and if compatible continue */
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struct call_leg *other = call_leg_other(&leg->base);
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if (!other) {
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LOGP(DSIP, LOGL_ERROR, "leg(%p) connected but leg gone\n", leg);
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nua_cancel(leg->nua_handle, TAG_END());
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return;
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}
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if (!extract_sdp(leg, sip)) {
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LOGP(DSIP, LOGL_ERROR, "leg(%p) incompatible audio, releasing\n", leg);
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nua_cancel(leg->nua_handle, TAG_END());
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other->release_call(other);
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return;
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}
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LOGP(DSIP, LOGL_NOTICE, "leg(%p) is now connected.\n", leg);
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leg->state = SIP_CC_CONNECTED;
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other->connect_call(other);
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nua_ack(leg->nua_handle, TAG_END());
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}
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void nua_callback(nua_event_t event, int status, char const *phrase, nua_t *nua, nua_magic_t *magic, nua_handle_t *nh, nua_hmagic_t *hmagic, sip_t const *sip, tagi_t tags[])
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{
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LOGP(DSIP, LOGL_DEBUG, "SIP event(%u) status(%d) phrase(%s) %p\n",
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event, status, phrase, hmagic);
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if (event == nua_r_invite) {
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struct sip_call_leg *leg;
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leg = (struct sip_call_leg *) hmagic;
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/* MT call is moving forward */
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/* The dialogue is now confirmed */
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if (leg->state == SIP_CC_INITIAL)
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leg->state = SIP_CC_DLG_CNFD;
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if (status == 180)
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call_progress(leg, sip);
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else if (status == 200)
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call_connect(leg, sip);
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else if (status >= 300) {
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struct call_leg *other = call_leg_other(&leg->base);
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LOGP(DSIP, LOGL_ERROR, "leg(%p) unknown err, releasing.\n", leg);
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nua_cancel(leg->nua_handle, TAG_END());
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nua_handle_destroy(leg->nua_handle);
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call_leg_release(&leg->base);
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if (other)
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other->release_call(other);
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}
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} else if (event == nua_r_bye || event == nua_r_cancel) {
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/* our bye or hang up is answered */
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struct sip_call_leg *leg = (struct sip_call_leg *) hmagic;
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LOGP(DSIP, LOGL_NOTICE, "leg(%p) got resp to %s\n",
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leg, event == nua_r_bye ? "bye" : "cancel");
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nua_handle_destroy(leg->nua_handle);
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call_leg_release(&leg->base);
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} else if (event == nua_i_bye) {
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/* our remote has hung up */
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struct sip_call_leg *leg = (struct sip_call_leg *) hmagic;
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struct call_leg *other = call_leg_other(&leg->base);
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LOGP(DSIP, LOGL_ERROR, "leg(%p) got bye, releasing.\n", leg);
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nua_handle_destroy(leg->nua_handle);
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call_leg_release(&leg->base);
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if (other)
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other->release_call(other);
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}
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}
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static void sip_release_call(struct call_leg *_leg)
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{
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struct sip_call_leg *leg;
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OSMO_ASSERT(_leg->type == CALL_TYPE_SIP);
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leg = (struct sip_call_leg *) _leg;
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/*
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* If a dialogue is not confirmed yet, we can probably not do much
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* but wait for the timeout. For a confirmed one we can send cancel
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* and for a connected one bye. I don't see how sofia-sip is going
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* to help us here.
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*/
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switch (leg->state) {
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case SIP_CC_INITIAL:
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LOGP(DSIP, LOGL_NOTICE, "Canceling leg(%p) in int state\n", leg);
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nua_handle_destroy(leg->nua_handle);
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call_leg_release(&leg->base);
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break;
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case SIP_CC_DLG_CNFD:
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LOGP(DSIP, LOGL_NOTICE, "Canceling leg(%p) in cnfd state\n", leg);
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nua_cancel(leg->nua_handle, TAG_END());
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break;
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case SIP_CC_CONNECTED:
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LOGP(DSIP, LOGL_NOTICE, "Ending leg(%p) in con\n", leg);
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nua_bye(leg->nua_handle, TAG_END());
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break;
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}
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}
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static const char *media_name(int ptmsg)
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{
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return "GSM";
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}
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static int send_invite(struct sip_agent *agent, struct sip_call_leg *leg,
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const char *calling_num, const char *called_num)
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{
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struct call_leg *other = leg->base.call->initial;
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struct in_addr net = { .s_addr = ntohl(other->ip) };
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leg->wanted_codec = media_name(other->payload_msg_type);
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char *from = talloc_asprintf(leg, "sip:%s@%s",
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calling_num,
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agent->app->sip.local_addr);
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char *to = talloc_asprintf(leg, "sip:%s@%s",
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called_num,
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agent->app->sip.remote_addr);
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char *sdp = talloc_asprintf(leg,
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"v=0\r\n"
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"o=Osmocom 0 0 IN IP4 %s\r\n"
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"s=GSM Call\r\n"
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"c=IN IP4 %s\r\n"
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"t=0 0\r\n"
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"m=audio %d RTP/AVP %d\r\n"
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"a=rtpmap:%d %s/8000\r\n",
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inet_ntoa(net), inet_ntoa(net), /* never use diff. addr! */
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other->port, other->payload_type,
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other->payload_type,
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leg->wanted_codec);
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leg->state = SIP_CC_INITIAL;
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nua_invite(leg->nua_handle,
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SIPTAG_FROM_STR(from),
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SIPTAG_TO_STR(to),
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NUTAG_MEDIA_ENABLE(0),
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SIPTAG_CONTENT_TYPE_STR("application/sdp"),
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SIPTAG_PAYLOAD_STR(sdp),
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TAG_END());
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leg->base.call->remote = &leg->base;
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talloc_free(from);
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talloc_free(to);
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talloc_free(sdp);
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return 0;
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}
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int sip_create_remote_leg(struct sip_agent *agent, struct call *call,
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const char *source, const char *dest)
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{
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struct sip_call_leg *leg;
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leg = talloc_zero(call, struct sip_call_leg);
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if (!leg) {
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LOGP(DSIP, LOGL_ERROR, "Failed to allocate leg for call(%u)\n",
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call->id);
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return -1;
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}
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leg->base.type = CALL_TYPE_SIP;
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leg->base.call = call;
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leg->base.release_call = sip_release_call;
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leg->agent = agent;
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leg->nua_handle = nua_handle(agent->nua, leg, TAG_END());
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if (!leg->nua_handle) {
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LOGP(DSIP, LOGL_ERROR, "Failed to allocate nua for call(%u)\n",
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call->id);
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talloc_free(leg);
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return -2;
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}
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return send_invite(agent, leg, source, dest);
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}
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char *make_sip_uri(struct sip_agent *agent)
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@ -8,6 +8,7 @@
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#include <sofia-sip/nua.h>
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struct app_config;
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struct call;
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struct sip_agent {
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struct app_config *app;
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@ -19,3 +20,5 @@ struct sip_agent {
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void sip_agent_init(struct sip_agent *agent, struct app_config *app);
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int sip_agent_start(struct sip_agent *agent);
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int sip_create_remote_leg(struct sip_agent *agent, struct call *call, const char *local, const char *remote);
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