@@ -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

-   <sr-users@lists.sip-router.org> and e-mails regarding development

-   versions should be sent to <sr-dev@lists.sip-router.org>.

+       E-mails regarding any stable Kamailio release should be sent to

+       <sr-users@lists.sip-router.org> and e-mails regarding development

+       versions should be sent to <sr-dev@lists.sip-router.org>.

-   If you want to keep the mail private, send it to

-   <sr-users@lists.sip-router.org>.

+       If you want to keep the mail private, send it to

+       <sr-users@lists.sip-router.org>.

    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.

@@ -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.