[svn-commits] file: branch file/pimp_sip_media r381010 - in /team/file/pimp_sip_media: conf...
SVN commits to the Digium repositories
svn-commits at lists.digium.com
Wed Feb 6 16:46:25 CST 2013
Author: file
Date: Wed Feb 6 16:46:21 2013
New Revision: 381010
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=381010
Log:
Incorporate review feedback from Mark.
Modified:
team/file/pimp_sip_media/configs/res_sip.conf.sample
team/file/pimp_sip_media/include/asterisk/res_sip_session.h
team/file/pimp_sip_media/res/res_sip_sdp_audio.c
team/file/pimp_sip_media/res/res_sip_session.c
Modified: team/file/pimp_sip_media/configs/res_sip.conf.sample
URL: http://svnview.digium.com/svn/asterisk/team/file/pimp_sip_media/configs/res_sip.conf.sample?view=diff&rev=381010&r1=381009&r2=381010
==============================================================================
--- team/file/pimp_sip_media/configs/res_sip.conf.sample (original)
+++ team/file/pimp_sip_media/configs/res_sip.conf.sample Wed Feb 6 16:46:21 2013
@@ -19,6 +19,6 @@
;timers_min_se=90 ; Minimum session timers expiration period, in seconds
;timers_sess_expires=1800 ; Session timers expiration period, in seconds
;mohsuggest=example ; What musiconhold class to suggest that the peer channel use when this endpoint places them on hold
-;rtp_ipv6=yes ; Use IPv6 for RTP transport
+;rtp_ipv6=yes ; Force IPv6 for RTP transport
;rtp_symmetric=yes ; Enable symmetric RTP support
;use_ptime=yes ; Whether to use the ptime value received from the endpoint or not
Modified: team/file/pimp_sip_media/include/asterisk/res_sip_session.h
URL: http://svnview.digium.com/svn/asterisk/team/file/pimp_sip_media/include/asterisk/res_sip_session.h?view=diff&rev=381010&r1=381009&r2=381010
==============================================================================
--- team/file/pimp_sip_media/include/asterisk/res_sip_session.h (original)
+++ team/file/pimp_sip_media/include/asterisk/res_sip_session.h Wed Feb 6 16:46:21 2013
@@ -84,8 +84,6 @@
struct ast_sip_session_media media[AST_SIP_MEDIA_SIZE];
/* Workspace for tasks relating to this SIP session */
struct ast_sip_work *work;
- /* Whether IPv6 RTP support is in use or not */
- unsigned int rtp_ipv6:1;
};
/*!
Modified: team/file/pimp_sip_media/res/res_sip_sdp_audio.c
URL: http://svnview.digium.com/svn/asterisk/team/file/pimp_sip_media/res/res_sip_sdp_audio.c?view=diff&rev=381010&r1=381009&r2=381010
==============================================================================
--- team/file/pimp_sip_media/res/res_sip_sdp_audio.c (original)
+++ team/file/pimp_sip_media/res/res_sip_sdp_audio.c Wed Feb 6 16:46:21 2013
@@ -81,7 +81,7 @@
char hostip[PJ_INET6_ADDRSTRLEN+2];
struct ast_sockaddr tmp;
- if (pj_gethostip(session->rtp_ipv6 ? pj_AF_INET6() : pj_AF_INET(), &addr) != PJ_SUCCESS) {
+ if (pj_gethostip(session->endpoint->rtp_ipv6 ? pj_AF_INET6() : pj_AF_INET(), &addr) != PJ_SUCCESS) {
return -1;
}
@@ -100,7 +100,7 @@
ast_rtp_codecs_packetization_set(ast_rtp_instance_get_codecs(session->media[AST_SIP_MEDIA_AUDIO].rtp),
session->media[AST_SIP_MEDIA_AUDIO].rtp, &session->endpoint->prefs);
-
+
return 0;
}
@@ -108,18 +108,14 @@
static int audio_negotiate_incoming_sdp_stream(struct ast_sip_session *session, const struct pjmedia_sdp_session *sdp, const struct pjmedia_sdp_media *stream)
{
char host[NI_MAXHOST];
- RAII_VAR(struct ast_sockaddr *, addrs, NULL, ast_free);
+ RAII_VAR(struct ast_sockaddr *, addrs, NULL, ast_free_ptr);
/* If no audio formats have been configured reject this stream */
if (!ast_format_cap_has_type(session->endpoint->codecs, AST_FORMAT_TYPE_AUDIO)) {
return 0;
}
- if (!stream->conn) {
- snprintf(host, sizeof(host), "%.*s", (int) pj_strlen(&sdp->conn->addr), pj_strbuf(&sdp->conn->addr));
- } else {
- snprintf(host, sizeof(host), "%.*s", (int) pj_strlen(&stream->conn->addr), pj_strbuf(&stream->conn->addr));
- }
+ ast_copy_pj_str(host, stream->conn ? &stream->conn->addr : &sdp->conn->addr, sizeof(host));
/* Ensure that the address provided is valid */
if (ast_sockaddr_resolve(&addrs, host, PARSE_PORT_FORBID, AST_AF_UNSPEC) <= 0) {
@@ -128,7 +124,7 @@
}
/* If IPv6 is configured but the media stream does not have it reject this stream */
- if (session->rtp_ipv6 && !ast_sockaddr_is_ipv6(addrs)) {
+ if (session->endpoint->rtp_ipv6 && !ast_sockaddr_is_ipv6(addrs)) {
return -1;
}
@@ -259,12 +255,14 @@
/*! \brief Function which applies a negotiated SDP stream */
static int audio_apply_negotiated_sdp_stream(struct ast_sip_session *session, const struct pjmedia_sdp_session *sdp, const struct pjmedia_sdp_media *stream)
{
- int res = 1, format, othercapability = 0;
+ int format, othercapability = 0;
char host[NI_MAXHOST];
- struct ast_sockaddr *addrs = NULL;
+ RAII_VAR(struct ast_sockaddr *, addrs, NULL, ast_free_ptr);
struct ast_rtp_codecs codecs;
const pjmedia_sdp_attr *attr;
- struct ast_format_cap *cap, *jointcap = NULL, *peercap = NULL;
+ RAII_VAR(struct ast_format_cap *, cap, NULL, ast_format_cap_destroy);
+ RAII_VAR(struct ast_format_cap *, jointcap, NULL, ast_format_cap_destroy);
+ RAII_VAR(struct ast_format_cap *, peercap, NULL, ast_format_cap_destroy);
struct ast_format fmt;
/* Create an RTP instance if need be */
@@ -272,11 +270,7 @@
return -1;
}
- if (!stream->conn) {
- snprintf(host, sizeof(host), "%.*s", (int) pj_strlen(&sdp->conn->addr), pj_strbuf(&sdp->conn->addr));
- } else {
- snprintf(host, sizeof(host), "%.*s", (int) pj_strlen(&stream->conn->addr), pj_strbuf(&stream->conn->addr));
- }
+ ast_copy_pj_str(host, stream->conn ? &stream->conn->addr : &sdp->conn->addr, sizeof(host));
/* Ensure that the address provided is valid */
if (ast_sockaddr_resolve(&addrs, host, PARSE_PORT_FORBID, AST_AF_UNSPEC) <= 0) {
@@ -288,8 +282,7 @@
if (!(cap = ast_format_cap_alloc_nolock()) ||
!(peercap = ast_format_cap_alloc_nolock())) {
ast_log(LOG_ERROR, "Failed to allocate audio capabilities\n");
- res = -1;
- goto cleanup;
+ return -1;
}
/* Apply connection information to the RTP instance */
@@ -311,7 +304,7 @@
if ((pjmedia_sdp_attr_to_rtpmap(session->inv_session->pool_active, attr, &rtpmap)) == PJ_SUCCESS) {
char name[32];
- snprintf(name, sizeof(name), "%.*s", (int) pj_strlen(&rtpmap->enc_name), pj_strbuf(&rtpmap->enc_name));
+ ast_copy_pj_str(name, &rtpmap->enc_name, sizeof(name));
ast_rtp_codecs_payloads_set_rtpmap_type_rate(&codecs, NULL, pj_strtoul(&stream->desc.fmt[format]),
"audio", name, 0, rtpmap->clock_rate);
}
@@ -353,8 +346,9 @@
ast_getformatname_multiple(usbuf, sizeof(usbuf), cap);
ast_getformatname_multiple(thembuf, sizeof(thembuf), peercap);
ast_log(LOG_WARNING, "No joint capabilities between our configuration(%s) and incoming SDP(%s)\n", usbuf, thembuf);
- res = -1;
- goto cleanup;
+
+ ast_rtp_codecs_payloads_destroy(&codecs);
+ return -1;
}
ast_rtp_codecs_payloads_copy(&codecs, ast_rtp_instance_get_codecs(session->media[AST_SIP_MEDIA_AUDIO].rtp),
@@ -394,14 +388,8 @@
ast_rtp_instance_activate(session->media[AST_SIP_MEDIA_AUDIO].rtp);
}
-cleanup:
- ast_format_cap_destroy(peercap);
- ast_format_cap_destroy(jointcap);
- ast_format_cap_destroy(cap);
- ast_free(addrs);
ast_rtp_codecs_payloads_destroy(&codecs);
-
- return res;
+ return 1;
}
/*! \brief Function which destroys the audio RTP instance when session ends */
Modified: team/file/pimp_sip_media/res/res_sip_session.c
URL: http://svnview.digium.com/svn/asterisk/team/file/pimp_sip_media/res/res_sip_session.c?view=diff&rev=381010&r1=381009&r2=381010
==============================================================================
--- team/file/pimp_sip_media/res/res_sip_session.c (original)
+++ team/file/pimp_sip_media/res/res_sip_session.c Wed Feb 6 16:46:21 2013
@@ -497,7 +497,6 @@
session->work = ast_sip_create_work();
session->endpoint = endpoint;
session->inv_session = inv_session;
- session->rtp_ipv6 = endpoint->rtp_ipv6;
if (add_supplements(session)) {
return NULL;
}
@@ -1083,6 +1082,7 @@
static const pj_str_t STR_ASTERISK = { "Asterisk", 8 };
static const pj_str_t STR_IN = { "IN", 2 };
static const pj_str_t STR_IP4 = { "IP4", 3 };
+ static const pj_str_t STR_IP6 = { "IP6", 3 };
pjmedia_sdp_session *local;
if (!(local = PJ_POOL_ZALLOC_T(inv->pool, pjmedia_sdp_session))) {
@@ -1098,10 +1098,7 @@
local->origin.user = STR_ASTERISK;
local->origin.net_type = STR_IN;
- local->origin.addr_type = STR_IP4;
- /* XXX Hardcoded origin net stuff for now. Replace with transport
- * stuff later
- */
+ local->origin.addr_type = session->endpoint->rtp_ipv6 ? STR_IP6 : STR_IP4;
local->origin.addr = *pj_gethostname();
local->name = local->origin.user;
More information about the svn-commits
mailing list