modules/rtpproxy: some README improvements

modules/rtpproxy/README

@@ -30,13 +30,13 @@ Carsten Bock
 
    ng-voice GmbH
 
-   Copyright © 2003-2008 Sippy Software, Inc.
+   Copyright © 2003-2008 Sippy Software, Inc.
 
-   Copyright © 2005 Voice Sistem SRL
+   Copyright © 2005 Voice Sistem SRL
 
-   Copyright © 2009 TuTPro Inc.
+   Copyright © 2009 TuTPro Inc.
 
-   Copyright © 2010 VoIPEmbedded Inc.
+   Copyright © 2010 VoIPEmbedded Inc.
      __________________________________________________________________
 
    Table of Contents
@@ -152,12 +152,10 @@ Chapter 1. Admin Guide
 1. Overview
 
    This is a module that enables media streams to be proxied via an
-   rtpproxy.
-
-   Known devices that get along over NATs with rtpproxy are ATAs (as
-   clients) and Cisco Gateways (since 12.2(T)) as servers. See
-   http://www.cisco.com/en/US/products/sw/iosswrel/ps1839/products_feature
-   _guide09186a0080110bf9.html">
+   rtpproxy. Rtpproxies know to work with this module are Sippy RTPproxy
+   http://www.rtpproxy.org and ngcp-rtpproxy-ng
+   http://deb.sipwise.com/spce/2.6/pool/main/n/ngcp-mediaproxy-ng. Some
+   features of rtpproxy module apply only one of the two rtpproxies.
 
 2. Multiple RTPProxy usage
 
@@ -168,7 +166,7 @@ Chapter 1. Admin Guide
    load-balancing will be performed over a set and the user has the
    ability to choose what set should be used. The set is selected via its
    id - the id being defined along with the set. Refer to the
-   "rtpproxy_sock" module parameter definition for syntax description.
+   “rtpproxy_sock� module parameter definition for syntax description.
 
    The balancing inside a set is done automatically by the module based on
    the weight of each rtpproxy from the set.
@@ -216,7 +214,7 @@ Chapter 1. Admin Guide
    Definition of socket(s) used to connect to (a set) RTPProxy. It may
    specify a UNIX socket or an IPv4/IPv6 UDP socket.
 
-   Default value is "NONE" (disabled).
+   Default value is “NONE� (disabled).
 
    Example 1.1. Set rtpproxy_sock parameter
 ...
@@ -238,7 +236,7 @@ modparam("rtpproxy", "rtpproxy_sock",
    will not attempt to establish communication to RTPProxy for
    rtpproxy_disable_tout seconds.
 
-   Default value is "60".
+   Default value is “60�.
 
    Example 1.2. Set rtpproxy_disable_tout parameter
 ...
@@ -249,7 +247,7 @@ modparam("rtpproxy", "rtpproxy_disable_tout", 20)
 
    Timeout value in waiting for reply from RTPProxy.
 
-   Default value is "1".
+   Default value is “1�.
 
    Example 1.3. Set rtpproxy_tout parameter
 ...
@@ -261,7 +259,7 @@ modparam("rtpproxy", "rtpproxy_tout", 2)
    How many times rtpproxy should retry to send and receive after timeout
    was generated.
 
-   Default value is "5".
+   Default value is “5�.
 
    Example 1.4. Set rtpproxy_retr parameter
 ...
@@ -273,7 +271,7 @@ modparam("rtpproxy", "rtpproxy_retr", 2)
    Socket to be forced in communicating to RTPProxy. It makes sense only
    for UDP communication. If no one specified, the OS will choose.
 
-   Default value is "NULL".
+   Default value is “NULL�.
 
    Example 1.5. Set force_socket parameter
 ...
@@ -291,7 +289,7 @@ Note
 
    The string must be a complete SDP line, including the EOH (\r\n).
 
-   Default value is "a=nortpproxy:yes\r\n".
+   Default value is “a=nortpproxy:yes\r\n�.
 
    Example 1.6. Set nortpproxy_str parameter
 ...
@@ -306,7 +304,7 @@ modparam("rtpproxy", "nortpproxy_str", "a=sdpmangled:yes\r\n")
    If it is an empty string, no timeout socket will be transmitted to the
    RTP-Proxy.
 
-   Default value is "" (nothing).
+   Default value is “� (nothing).
 
    Example 1.7. Set timeout_socket parameter
 ...
@@ -327,7 +325,7 @@ modparam("nathelper", "timeout_socket", "xmlrpc:http://127.0.0.1:8000/RPC2")
    5.10. start_recording()
    5.11. rtpproxy_stop_stream2uas(prompt_name, count)
 
-5.1. set_rtp_proxy_set(setid)
+5.1.  set_rtp_proxy_set(setid)
 
    Sets the Id of the rtpproxy set to be used for the next
    unforce_rtp_proxy(), rtpproxy_offer(), rtpproxy_answer() or
@@ -343,7 +341,7 @@ set_rtp_proxy_set("2");
 rtpproxy_offer();
 ...
 
-5.2. rtpproxy_offer([flags [, ip_address]])
+5.2.  rtpproxy_offer([flags [, ip_address]])
 
    Rewrites SDP body to ensure that media is passed through an RTP proxy.
    To be invoked on INVITE for the cases the SDPs are in INVITE and 200 OK
@@ -353,16 +351,16 @@ rtpproxy_offer();
      * flags - flags to turn on some features.
           + 1 - append first Via branch to Call-ID when sending command to
             rtpproxy. This can be used to create one media session per
-            branch on the rtpproxy. When sending a subsequent "delete"
+            branch on the rtpproxy. When sending a subsequent “delete�
             command to the rtpproxy, you can then stop just the session
             for a specific branch when passing the flag '1' or '2' in the
-            "unforce_rtpproxy", or stop all sessions for a call when not
+            “unforce_rtpproxy�, or stop all sessions for a call when not
             passing one of those two flags there. This is especially
             useful if you have serially forked call scenarios where
-            rtpproxy gets an "update" command for a new branch, and then a
-            "delete" command for the previous branch, which would
+            rtpproxy gets an “update� command for a new branch, and then a
+            “delete� command for the previous branch, which would
             otherwise delete the full call, breaking the subsequent
-            "lookup" for the new branch. This flag is only supported by
+            “lookup� for the new branch. This flag is only supported by
             the ngcp-mediaproxy-ng rtpproxy at the moment!
           + 2 - append second Via branch to Call-ID when sending command
             to rtpproxy. See flag '1' for its meaning.
@@ -370,7 +368,7 @@ rtpproxy_offer();
             set for a reply.
           + a - flags that UA from which message is received doesn't
             support symmetric RTP. (automatically sets the 'r' flag)
-          + l - force "lookup", that is, only rewrite SDP when
+          + l - force “lookup�, that is, only rewrite SDP when
             corresponding session is already exists in the RTP proxy. By
             default is on when the session is to be completed.
           + i, e - these flags specify the direction of the SIP message.
@@ -454,7 +452,7 @@ onreply_route[2]
 ...
 }
 
-5.3. rtpproxy_answer([flags [, ip_address]])
+5.3.  rtpproxy_answer([flags [, ip_address]])
 
    Rewrites SDP body to ensure that media is passed through an RTP proxy.
    To be invoked on 200 OK for the cases the SDPs are in INVITE and 200 OK
@@ -470,7 +468,7 @@ onreply_route[2]
 
    See rtpproxy_offer() function example above for example.
 
-5.4. rtpproxy_destroy([flags])
+5.4.  rtpproxy_destroy([flags])
 
    Tears down the RTPProxy session for the current call.
 
@@ -480,16 +478,16 @@ onreply_route[2]
      * flags - flags to turn on some features.
           + 1 - append first Via branch to Call-ID when sending command to
             rtpproxy. This can be used to create one media session per
-            branch on the rtpproxy. When sending a subsequent "delete"
+            branch on the rtpproxy. When sending a subsequent “delete�
             command to the rtpproxy, you can then stop just the session
             for a specific branch when passing the flag '1' or '2' in the
-            "unforce_rtpproxy", or stop all sessions for a call when not
+            “unforce_rtpproxy�, or stop all sessions for a call when not
             passing one of those two flags there. This is especially
             useful if you have serially forked call scenarios where
-            rtpproxy gets an "update" command for a new branch, and then a
-            "delete" command for the previous branch, which would
+            rtpproxy gets an “update� command for a new branch, and then a
+            “delete� command for the previous branch, which would
             otherwise delete the full call, breaking the subsequent
-            "lookup" for the new branch. This flag is only supported by
+            “lookup� for the new branch. This flag is only supported by
             the ngcp-mediaproxy-ng rtpproxy at the moment!
           + 2 - append second Via branch to Call-ID when sending command
             to rtpproxy. See flag '1' for its meaning.
@@ -499,11 +497,11 @@ onreply_route[2]
 rtpproxy_destroy();
 ...
 
-5.5. unforce_rtp_proxy()
+5.5.  unforce_rtp_proxy()
 
    Same as rtpproxy_destroy().
 
-5.6. rtpproxy_manage([flags [, ip_address]])
+5.6.  rtpproxy_manage([flags [, ip_address]])
 
    Manage the RTPProxy session - it combines the functionality of
    rtpproxy_offer(), rtpproxy_answer() and unfroce_rtpproxy(), detecting
@@ -531,7 +529,7 @@ rtpproxy_destroy();
 rtpproxy_manage();
 ...
 
-5.7. rtpproxy_stream2uac(prompt_name, count),
+5.7.  rtpproxy_stream2uac(prompt_name, count),
 
    Instruct the RTPproxy to stream prompt/announcement pre-encoded with
    the makeann command from the RTPproxy distribution. The uac/uas suffix
@@ -573,11 +571,11 @@ rtpproxy_manage();
     };
 ...
 
-5.8. rtpproxy_stream2uas(prompt_name, count)
+5.8.  rtpproxy_stream2uas(prompt_name, count)
 
    See function rtpproxy_stream2uac(prompt_name, count).
 
-5.9. rtpproxy_stop_stream2uac(),
+5.9.  rtpproxy_stop_stream2uac(),
 
    Stop streaming of announcement/prompt/MOH started previously by the
    respective rtpproxy_stream2xxx. The uac/uas suffix selects whose
@@ -586,10 +584,11 @@ rtpproxy_manage();
 
    These functions can be used from REQUEST_ROUTE, ONREPLY_ROUTE.
 
-5.10. start_recording()
+5.10.  start_recording()
 
-   This command will send a signal to the RTP-Proxy to record the RTP
-   stream on the RTP-Proxy.
+   This function will send a signal to the RTP-Proxy to record the RTP
+   stream on the RTP-Proxy. This function is only supported by Sippy
+   RTPproxy at the moment!
 
    This function can be used from REQUEST_ROUTE and ONREPLY_ROUTE.
 
@@ -598,7 +597,7 @@ rtpproxy_manage();
 start_recording();
 ...
 
-5.11. rtpproxy_stop_stream2uas(prompt_name, count)
+5.11.  rtpproxy_stop_stream2uas(prompt_name, count)
 
    See function rtpproxy_stop_stream2uac(prompt_name, count).
 
@@ -636,7 +635,7 @@ start_recording();
    NOTE: if a rtpproxy is defined multiple times (in the same or diferente
    sete), all its instances will be enables/disabled.
 
-   Example 1.16. nh_enable_rtpp usage
+   Example 1.16.  nh_enable_rtpp usage
 ...
 $ kamctl fifo nh_enable_rtpp udp:192.168.2.133:8081 0
 ...
@@ -648,52 +647,52 @@ $ kamctl fifo nh_enable_rtpp udp:192.168.2.133:8081 0
 
    No parameter.
 
-   Example 1.17. nh_show_rtpp usage
+   Example 1.17.  nh_show_rtpp usage
 ...
 $ kamctl fifo nh_show_rtpp
 ...
 
 Chapter 2. Frequently Asked Questions
 
-   2.1. What happend with "rtpproxy_disable" parameter?
+   2.1. What happend with “rtpproxy_disable� parameter?
    2.2. Where can I find more about Kamailio?
    2.3. Where can I post a question about this module?
    2.4. How can I report a bug?
 
    2.1.
 
-   What happend with "rtpproxy_disable" parameter?
+       What happend with “rtpproxy_disable� parameter?
 
-   It was removed as it became obsolete - now "rtpproxy_sock" can take
-   empty value to disable the rtpproxy functionality.
+       It was removed as it became obsolete - now “rtpproxy_sock� can take
+       empty value to disable the rtpproxy functionality.
 
    2.2.
 
-   Where can I find more about Kamailio?
+       Where can I find more about Kamailio?
 
-   Take a look at http://www.kamailio.org/.
+       Take a look at http://www.kamailio.org/.
 
    2.3.
 
-   Where can I post a question about this module?
+       Where can I post a question about this module?
 
-   First at all check if your question was already answered on one of our
-   mailing lists:
-     * User Mailing List -
-       http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
-     * Developer Mailing List -
-       http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-dev
+       First at all check if your question was already answered on one of our
+       mailing lists:
+         * User Mailing List -
+           http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
+         * Developer Mailing List -
+           http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-dev
 
-   E-mails regarding any stable Kamailio release should be sent to
-   <[email protected]> and e-mails regarding development
-   versions should be sent to <[email protected]>.
+       E-mails regarding any stable Kamailio release should be sent to
+       <[email protected]> and e-mails regarding development
+       versions should be sent to <[email protected]>.
 
-   If you want to keep the mail private, send it to
-   <[email protected]>.
+       If you want to keep the mail private, send it to
+       <[email protected]>.
 
    2.4.
 
-   How can I report a bug?
+       How can I report a bug?
 
-   Please follow the guidelines provided at:
-   http://sip-router.org/tracker.
+       Please follow the guidelines provided at:
+       http://sip-router.org/tracker.

modules/rtpproxy/doc/rtpproxy_admin.xml

@@ -18,13 +18,9 @@
 	<title>Overview</title>
 	<para>
 		This is a module that enables media streams to be proxied
-		via an rtpproxy.
-	</para>
-	<para>
-		Known devices that get along over &nat;s with rtpproxy are ATAs
-		(as clients) and Cisco Gateways (since 12.2(T)) as servers.  See <ulink
-		url="http://www.cisco.com/en/US/products/sw/iosswrel/ps1839/products_feature_guide09186a0080110bf9.html">
-		http://www.cisco.com/en/US/products/sw/iosswrel/ps1839/products_feature_guide09186a0080110bf9.html"></ulink>
+		via an rtpproxy.  Rtpproxies know to work with this module
+		are Sippy RTPproxy <ulink url="http://www.rtpproxy.org"></ulink>
+		and ngcp-rtpproxy-ng <ulink url="http://deb.sipwise.com/spce/2.6/pool/main/n/ngcp-mediaproxy-ng"></ulink>.  Some features of rtpproxy module apply only one of the two rtpproxies.
 	</para>
 	</section>
 
@@ -673,8 +669,8 @@ rtpproxy_manage();
 		<function moreinfo="none">start_recording()</function>
 		</title>
 		<para>
-		This command will send a signal to the RTP-Proxy to record
-		the RTP stream on the RTP-Proxy.
+		This function will send a signal to the RTP-Proxy to record
+		the RTP stream on the RTP-Proxy. <emphasis>This function is only supported by Sippy RTPproxy at the moment!</emphasis>
 		</para>
 		<para>
 		This function can be used from REQUEST_ROUTE and ONREPLY_ROUTE.