sip/sdp: Hack to get a GSM codec call connected

Extract IP and port once we have received the INVITE. In the near
future we need to improve this!

3 files changed, 19 insertions, 4 deletions

diff --git a/src/sdp.c b/src/sdp.c
index 2193ab9..ccd000d 100644
--- a/src/sdp.c
+++ b/src/sdp.c
@@ -90,7 +90,7 @@ success:
 	return true;
 }
 
-bool sdp_extract_sdp(struct sip_call_leg *leg, const sip_t *sip)
+bool sdp_extract_sdp(struct sip_call_leg *leg, const sip_t *sip, bool any_codec)
 {
 	sdp_connection_t *conn;
 	sdp_session_t *sdp;
@@ -139,7 +139,7 @@ bool sdp_extract_sdp(struct sip_call_leg *leg, const sip_t *sip)
 			continue;
 
 		for (map = media->m_rtpmaps; map; map = map->rm_next) {
-			if (strcasecmp(map->rm_encoding, leg->wanted_codec) != 0)
+			if (!any_codec && strcasecmp(map->rm_encoding, leg->wanted_codec) != 0)
 				continue;
 
 			leg->base.port = media->m_port;
diff --git a/src/sdp.h b/src/sdp.h
index aad736d..d716644 100644
--- a/src/sdp.h
+++ b/src/sdp.h
@@ -8,6 +8,6 @@ struct sip_call_leg;
 struct call_leg;
 
 bool sdp_screen_sdp(const sip_t *sip);
-bool sdp_extract_sdp(struct sip_call_leg *leg, const sip_t *sip);
+bool sdp_extract_sdp(struct sip_call_leg *leg, const sip_t *sip, bool any_codec);
 
 char *sdp_create_file(struct sip_call_leg *, struct call_leg *);
diff --git a/src/sip.c b/src/sip.c
index 5c2b5ab..36e4967 100644
--- a/src/sip.c
+++ b/src/sip.c
@@ -60,7 +60,7 @@ static void call_connect(struct sip_call_leg *leg, const sip_t *sip)
 		return;
 	}
 
-	if (!sdp_extract_sdp(leg, sip)) {
+	if (!sdp_extract_sdp(leg, sip, false)) {
 		LOGP(DSIP, LOGL_ERROR, "leg(%p) incompatible audio, releasing\n", leg);
 		nua_cancel(leg->nua_handle, TAG_END());
 		other->release_call(other);
@@ -113,6 +113,21 @@ static void new_call(struct sip_agent *agent, nua_handle_t *nh,
 	leg->state = SIP_CC_DLG_CNFD;
 	leg->dir = SIP_DIR_MO;
 
+	/*
+	 * FIXME/TODO.. we need to select the codec at some point. But it is
+	 * not this place. It starts with the TCH/F vs. TCH/H selection based
+	 * on the offered codecs, and then RTP_CREATE should have it. So both
+	 * are GSM related... and do not belong here. Just pick the first codec
+	 * so the IP addresss port and payload type is set.
+	 */
+	if (!sdp_extract_sdp(leg, sip, true)) {
+		LOGP(DSIP, LOGL_ERROR, "leg(%p) no audio, releasing\n", leg);
+		nua_respond(nh, SIP_406_NOT_ACCEPTABLE, TAG_END());
+		nua_handle_destroy(nh);
+		call_leg_release(&leg->base);
+		return;
+	}
+
 	leg->base.release_call = sip_release_call;
 	leg->base.ring_call = sip_ring_call;
 	leg->base.connect_call = sip_connect_call;