-
rtpproxy Module
Maxim Sobolev
Copyright © 2009-2012 TuTPro Inc.
Copyright © 2010 VoIPEmbedded Inc.
- _________________________________________________________________
+ __________________________________________________________________
Table of Contents
5. Functions
- 5.1. set_rtp_proxy_set(setid)
- 5.2. rtpproxy_offer([flags [, ip_address]])
- 5.3. rtpproxy_answer([flags [, ip_address]])
- 5.4. rtpproxy_destroy([flags])
- 5.5. unforce_rtp_proxy()
- 5.6. rtpproxy_manage([flags [, ip_address]])
- 5.7. rtpproxy_stream2uac(prompt_name, count),
- 5.8. rtpproxy_stream2uas(prompt_name, count)
- 5.9. rtpproxy_stop_stream2uac(),
- 5.10. start_recording()
- 5.11. rtpproxy_stop_stream2uas(prompt_name, count)
+ 5.1. set_rtp_proxy_set(setid)
+ 5.2. rtpproxy_offer([flags [, ip_address]])
+ 5.3. rtpproxy_answer([flags [, ip_address]])
+ 5.4. rtpproxy_destroy([flags])
+ 5.5. unforce_rtp_proxy()
+ 5.6. rtpproxy_manage([flags [, ip_address]])
+ 5.7. rtpproxy_stream2uac(prompt_name, count),
+ 5.8. rtpproxy_stream2uas(prompt_name, count)
+ 5.9. rtpproxy_stop_stream2uac(),
+ 5.10. start_recording()
+ 5.11. rtpproxy_stop_stream2uas(prompt_name, count)
6. Exported Pseudo Variables
5. Functions
- 5.1. set_rtp_proxy_set(setid)
- 5.2. rtpproxy_offer([flags [, ip_address]])
- 5.3. rtpproxy_answer([flags [, ip_address]])
- 5.4. rtpproxy_destroy([flags])
- 5.5. unforce_rtp_proxy()
- 5.6. rtpproxy_manage([flags [, ip_address]])
- 5.7. rtpproxy_stream2uac(prompt_name, count),
- 5.8. rtpproxy_stream2uas(prompt_name, count)
- 5.9. rtpproxy_stop_stream2uac(),
- 5.10. start_recording()
- 5.11. rtpproxy_stop_stream2uas(prompt_name, count)
+ 5.1. set_rtp_proxy_set(setid)
+ 5.2. rtpproxy_offer([flags [, ip_address]])
+ 5.3. rtpproxy_answer([flags [, ip_address]])
+ 5.4. rtpproxy_destroy([flags])
+ 5.5. unforce_rtp_proxy()
+ 5.6. rtpproxy_manage([flags [, ip_address]])
+ 5.7. rtpproxy_stream2uac(prompt_name, count),
+ 5.8. rtpproxy_stream2uas(prompt_name, count)
+ 5.9. rtpproxy_stop_stream2uac(),
+ 5.10. start_recording()
+ 5.11. rtpproxy_stop_stream2uas(prompt_name, count)
6. Exported Pseudo Variables
1. Overview
- This is a module that enables media streams to be proxied via an
- rtpproxy. Rtpproxies know to work with this module are Sippy RTPproxy
+ This is a module that enables media streams to be proxied via an
+ 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 the rtpproxy module apply only to one of the two
+ http://deb.sipwise.com/spce/2.6/pool/main/n/ngcp-mediaproxy-ng. Some
+ features of the rtpproxy module apply only to one of the two
rtpproxies.
2. Multiple RTPProxy usage
- The rtpproxy module can support multiple rtpproxies for
+ The rtpproxy module can support multiple rtpproxies for
balancing/distribution and control/selection purposes.
- The module allows definition of several sets of rtpproxies.
- Load-balancing will be performed over a set and the admin has the
+ The module allows definition of several sets of rtpproxies.
+ Load-balancing will be performed over a set and the admin has the
ability to choose what set should be used. The set is selected via its
- id - the id being defined with the set. Refer to the "rtpproxy_sock"
+ id - the id being defined with the set. Refer to the "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.
+ The balancing inside a set is done automatically by the module based on
+ the weight of each rtpproxy from the set.
- The selection of the set is done from script prior using
+ The selection of the set is done from script prior using
unforce_rtp_proxy(), rtpproxy_offer() or rtpproxy_answer() functions -
see the set_rtp_proxy_set() function.
- For backward compatibility reasons, a set with no id take by default
+ For backward compatibility reasons, a set with no id take by default
the id 0. Also if no set is explicitly set before unforce_rtp_proxy(),
rtpproxy_offer() or rtpproxy_answer() the 0 id set will be used.
- IMPORTANT: if you use multiple sets, take care and use the same set
- for both rtpproxy_offer()/rtpproxy_answer() and unforce_rtpproxy()!!
+ IMPORTANT: if you use multiple sets, take care and use the same set for
+ both rtpproxy_offer()/rtpproxy_answer() and unforce_rtpproxy()!!
3. Dependencies
3.2. External Libraries or Applications
- The following libraries or applications must be installed before
+ The following libraries or applications must be installed before
running Kamailio with this module loaded:
* None.
4.1. rtpproxy_sock (string)
Used to define the list of RTPPRoxy instances to connect to. These can
- be UNIX sockets or IPv4/IPv6 UDP sockets. Each modparam entry will
- insert sockets into a single set. If no set ID is given, the default
+ be UNIX sockets or IPv4/IPv6 UDP sockets. Each modparam entry will
+ insert sockets into a single set. If no set ID is given, the default
set ID '0' will be used. To define multiple sets add the set number at
- the beginning of each parameter followed by '=='. Sockets can be
- weighted by adding '=#' to a socket where # is an integer. A socket
- with a weight of 2 will be chosen twice as often as one with a weight
+ the beginning of each parameter followed by '=='. Sockets can be
+ weighted by adding '=#' to a socket where # is an integer. A socket
+ with a weight of 2 will be chosen twice as often as one with a weight
of 1.
- Default value is "NONE" (disabled).
+ Default value is "NONE" (disabled).
Example 1.1. Set rtpproxy_sock parameter
...
4.2. rtpproxy_disable_tout (integer)
- Once RTPProxy was found unreachable and marked as disabled, the
- rtpproxy module will not attempt to establish communication to
- RTPProxy for rtpproxy_disable_tout seconds.
+ Once RTPProxy was found unreachable and marked as disabled, the
+ rtpproxy module 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
...
Timeout value in waiting for reply from RTPProxy.
- Default value is "1".
+ Default value is "1".
Example 1.3. Set rtpproxy_tout parameter
...
4.4. rtpproxy_retr (integer)
- How many times the module should retry to send and receive after
+ How many times the module 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
...
4.5. nortpproxy_str (string)
- This parameter sets the SDP attribute used by rtpproxy to mark the
- message's SDP attachemnt with information that it have already been
+ This parameter sets the SDP attribute used by rtpproxy to mark the
+ message's SDP attachemnt with information that it have already been
changed.
If empty string, no marker will be added or checked.
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 1.5. Set nortpproxy_str parameter
...
4.6. timeout_socket (string)
The parameter sets the RTP timeout socket, which is transmitted to the
- RTP-Proxy. It will be used by the RTP proxy to signal back that a
- media stream timed out.
+ RTP-Proxy. It will be used by the RTP proxy to signal back that a media
+ stream timed out.
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 1.6. Set timeout_socket parameter
...
4.7. ice_candidate_priority_avp (string)
- If specified and if value of the avp value is not 0, rtpproxy_manage
- function adds ICE relay candidate attributes to sdp stream(s)
+ If specified and if value of the avp value is not 0, rtpproxy_manage
+ function adds ICE relay candidate attributes to sdp stream(s)
containing ICE candidate attributes.
- If value of the avp is 1, added candidates have high priority. If
- value of the avp is 2 (default), added candidates have low priority.
+ If value of the avp is 1, added candidates have high priority. If value
+ of the avp is 2 (default), added candidates have low priority.
- There is no default value meaning that no ICE relay candidates are
- added in any circumstance.
+ There is no default value meaning that no ICE relay candidates are
+ added in any circumstance.
Example 1.7. Set ice_candidate_priority_avp parameter
...
4.8. extra_id_pv (string)
- The parameter sets the PV defination to use when the "b" parameter is
- used on unforce_rtp_proxy(), rtpproxy_offer(), rtpproxy_answer() or
+ The parameter sets the PV defination to use when the "b" parameter is
+ used on unforce_rtp_proxy(), rtpproxy_offer(), rtpproxy_answer() or
rtpproxy_manage() command.
Default is empty, the "b" parameter may not be used then.
4.9. db_url (string)
- The database URL to load rtp_proxy sets from. If this parameter is
- set, the module will attempt to load the rtpproxy sets from the
- specified database and will ignore any 'rtpproxy_sock' modparams.
+ The database URL to load rtp_proxy sets from. If this parameter is set,
+ the module will attempt to load the rtpproxy sets from the specified
+ database and will ignore any 'rtpproxy_sock' modparams.
Default is empty, a database will not be used.
4.11. rtp_inst_pvar (string)
- A pseudo variable to store the chosen RTPProxy address. If this
- parameter is set, the instance URL will be stored in the given
+ A pseudo variable to store the chosen RTPProxy address. If this
+ parameter is set, the instance URL will be stored in the given
variable.
By default, this parameter is not set.
5. Functions
- 5.1. set_rtp_proxy_set(setid)
- 5.2. rtpproxy_offer([flags [, ip_address]])
- 5.3. rtpproxy_answer([flags [, ip_address]])
- 5.4. rtpproxy_destroy([flags])
- 5.5. unforce_rtp_proxy()
- 5.6. rtpproxy_manage([flags [, ip_address]])
- 5.7. rtpproxy_stream2uac(prompt_name, count),
- 5.8. rtpproxy_stream2uas(prompt_name, count)
- 5.9. rtpproxy_stop_stream2uac(),
- 5.10. start_recording()
- 5.11. rtpproxy_stop_stream2uas(prompt_name, count)
-
-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
+ 5.1. set_rtp_proxy_set(setid)
+ 5.2. rtpproxy_offer([flags [, ip_address]])
+ 5.3. rtpproxy_answer([flags [, ip_address]])
+ 5.4. rtpproxy_destroy([flags])
+ 5.5. unforce_rtp_proxy()
+ 5.6. rtpproxy_manage([flags [, ip_address]])
+ 5.7. rtpproxy_stream2uac(prompt_name, count),
+ 5.8. rtpproxy_stream2uas(prompt_name, count)
+ 5.9. rtpproxy_stop_stream2uac(),
+ 5.10. start_recording()
+ 5.11. rtpproxy_stop_stream2uas(prompt_name, count)
+
+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
rtpproxy_manage() command. The parameter can be an integer or a config
variable holding an integer.
- This function can be used from REQUEST_ROUTE, ONREPLY_ROUTE,
+ This function can be used from REQUEST_ROUTE, ONREPLY_ROUTE,
BRANCH_ROUTE.
Example 1.13. set_rtp_proxy_set usage
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 and on 200 OK when SDPs are in 200 OK and ACK.
+ To be invoked on INVITE for the cases the SDPs are in INVITE and 200 OK
+ and on 200 OK when SDPs are in 200 OK and ACK.
Meaning of the parameters is as follows:
* 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"
- command to the rtpproxy, you can then stop just the session
+ + 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"
+ 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
- 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
- otherwise delete the full call, breaking the subsequent
- "lookup" for the new branch. This flag is only supported by
+ "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
+ otherwise delete the full call, breaking the subsequent
+ "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
+ + 2 - append second Via branch to Call-ID when sending command
to rtpproxy. See flag '1' for its meaning.
- + 3 - behave like flag 1 is set for a request and like flag 2
- is set for a reply.
- + a - flags that UA from which message is received doesn't
+ + 3 - behave like flag 1 is set for a request and like flag 2 is
+ set for a reply.
+ + a - flags that UA from which message is received doesn't
support symmetric RTP. (automatically sets the 'r' flag)
- + b - append branch specific variable to Call-ID when sending
- command to rtpproxy. This creates one rtpproxy session per
- unique variable. Works similar to the 1, 2 and 3 parameter,
- but is usefull when forking to multiple destinations on
- different address families or network segments, requiring
- different rtpproxy parameters. The variable value is taken
- from the "extra_id_pv". When used, it must be used in every
- call to rtpproxy_manage(), rtpproxy_offer(),
- rtpproxy_answer() and rtpproxy_destroy() with the same
- contents of the PV. The b parameter may not be used in
- conjunction with the 1, 2 or 3 parameter to use the Via
- branch in the Call-ID.
- + l - force "lookup", that is, only rewrite SDP when
- corresponding session already exists in the RTP proxy. By
+ + b - append branch specific variable to Call-ID when sending
+ command to rtpproxy. This creates one rtpproxy session per
+ unique variable. Works similar to the 1, 2 and 3 parameter,
+ but is usefull when forking to multiple destinations on
+ different address families or network segments, requiring
+ different rtpproxy parameters. The variable value is taken
+ from the "extra_id_pv". When used, it must be used in every
+ call to rtpproxy_manage(), rtpproxy_offer(), rtpproxy_answer()
+ and rtpproxy_destroy() with the same contents of the PV. The b
+ parameter may not be used in conjunction with the 1, 2 or 3
+ parameter to use the Via branch in the Call-ID.
+ + l - force "lookup", that is, only rewrite SDP when
+ corresponding session 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.
- These flags only make sense when rtpproxy is running in
- bridge mode. 'i' means internal network (LAN), 'e' means
- external network (WAN). 'i' corresponds to rtpproxy's first
- interface, 'e' corresponds to rtpproxy's second interface.
- You always have to specify two flags to define the incoming
- network and the outgoing network. For example, 'ie' should be
- used for SIP message received from the local interface and
- sent out on the external interface, and 'ei' vice versa.
- Other options are 'ii' and 'ee'. So, for example if a SIP
- requests is processed with 'ie' flags, the corresponding
- response must be processed with 'ie' flags.
- Note: As rtpproxy in bridge mode s per default asymmetric,
- you have to specify the 'w' flag for clients behind NAT! See
- also above notes!
- + x - this flag a shortcut for using the "ie" or "ei"-flags of
- RTP-Proxy, in order to do automatic bridging between IPv4 on
- the "internal network" and IPv6 on the "external network".
- The distinction is done by the given IP in the SDP, e.g. a
- IPv4 Address will always call "ie" to the RTPProxy (IPv4(i)
- to IPv6(e)) and an IPv6Address will always call "ei" to the
+ + i, e - these flags specify the direction of the SIP message.
+ These flags only make sense when rtpproxy is running in bridge
+ mode. 'i' means internal network (LAN), 'e' means external
+ network (WAN). 'i' corresponds to rtpproxy's first interface,
+ 'e' corresponds to rtpproxy's second interface. You always
+ have to specify two flags to define the incoming network and
+ the outgoing network. For example, 'ie' should be used for SIP
+ message received from the local interface and sent out on the
+ external interface, and 'ei' vice versa. Other options are
+ 'ii' and 'ee'. So, for example if a SIP requests is processed
+ with 'ie' flags, the corresponding response must be processed
+ with 'ie' flags.
+ Note: As rtpproxy in bridge mode s per default asymmetric, you
+ have to specify the 'w' flag for clients behind NAT! See also
+ above notes!
+ + x - this flag a shortcut for using the "ie" or "ei"-flags of
+ RTP-Proxy, in order to do automatic bridging between IPv4 on
+ the "internal network" and IPv6 on the "external network". The
+ distinction is done by the given IP in the SDP, e.g. a IPv4
+ Address will always call "ie" to the RTPProxy (IPv4(i) to
+ IPv6(e)) and an IPv6Address will always call "ei" to the
RTPProxy (IPv6(e) to IPv4(i)).
- Note: Please note, that this will only work properly with
- non-dual-stack user-agents or with dual-stack clients
- according to RFC6157 (which suggest ICE for Dual-Stack
- implementations). This short-cut will not work properly with
- RFC4091 (ANAT) compatible clients, which suggests having
- different m-lines with different IP-protocols grouped
+ Note: Please note, that this will only work properly with
+ non-dual-stack user-agents or with dual-stack clients
+ according to RFC6157 (which suggest ICE for Dual-Stack
+ implementations). This short-cut will not work properly with
+ RFC4091 (ANAT) compatible clients, which suggests having
+ different m-lines with different IP-protocols grouped
together.
- + f - instructs rtpproxy to ignore marks inserted by another
- rtpproxy in transit to indicate that the session is already
- goes through another proxy. Allows creating a chain of
+ + f - instructs rtpproxy to ignore marks inserted by another
+ rtpproxy in transit to indicate that the session is already
+ goes through another proxy. Allows creating a chain of
proxies.
- + r - flags that IP address in SDP should be trusted. Without
- this flag, rtpproxy ignores address in the SDP and uses
- source address of the SIP message as media address which is
- passed to the RTP proxy.
- + o - flags that IP from the origin description (o=) should be
+ + r - flags that IP address in SDP should be trusted. Without
+ this flag, rtpproxy ignores address in the SDP and uses source
+ address of the SIP message as media address which is passed to
+ the RTP proxy.
+ + o - flags that IP from the origin description (o=) should be
also changed.
- + c - flags to change the session-level SDP connection (c=) IP
+ + c - flags to change the session-level SDP connection (c=) IP
if media-description also includes connection information.
- + w - flags that for the UA from which message is received,
+ + w - flags that for the UA from which message is received,
support symmetric RTP must be forced.
- + zNN - requests the RTPproxy to perform re-packetization of
- RTP traffic coming from the UA which has sent the current
- message to increase or decrease payload size per each RTP
- packet forwarded if possible. The NN is the target payload
- size in ms, for the most codecs its value should be in 10ms
- increments, however for some codecs the increment could
- differ (e.g. 30ms for GSM or 20ms for G.723). The RTPproxy
- would select the closest value supported by the codec. This
- feature could be used for significantly reducing bandwith
- overhead for low bitrate codecs, for example with G.729 going
- from 10ms to 100ms saves two thirds of the network bandwith.
+ + zNN - requests the RTPproxy to perform re-packetization of RTP
+ traffic coming from the UA which has sent the current message
+ to increase or decrease payload size per each RTP packet
+ forwarded if possible. The NN is the target payload size in
+ ms, for the most codecs its value should be in 10ms
+ increments, however for some codecs the increment could differ
+ (e.g. 30ms for GSM or 20ms for G.723). The RTPproxy would
+ select the closest value supported by the codec. This feature
+ could be used for significantly reducing bandwith overhead for
+ low bitrate codecs, for example with G.729 going from 10ms to
+ 100ms saves two thirds of the network bandwith.
* ip_address - new SDP IP address.
This function can be used from ANY_ROUTE.
...
}
-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 and on ACK when SDPs are in 200 OK and ACK.
+ To be invoked on 200 OK for the cases the SDPs are in INVITE and 200 OK
+ and on ACK when SDPs are in 200 OK and ACK.
- See rtpproxy_answer() function description above for the meaning of
- the parameters.
+ See rtpproxy_answer() function description above for the meaning of the
+ parameters.
- This function can be used from REQUEST_ROUTE, ONREPLY_ROUTE,
+ This function can be used from REQUEST_ROUTE, ONREPLY_ROUTE,
FAILURE_ROUTE, BRANCH_ROUTE.
Example 1.15. rtpproxy_answer usage
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.
Meaning of the parameters is as follows:
* 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"
- command to the rtpproxy, you can then stop just the session
+ + 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"
+ 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
- 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
- otherwise delete the full call, breaking the subsequent
- "lookup" for the new branch. This flag is only supported by
+ "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
+ otherwise delete the full call, breaking the subsequent
+ "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
+ + 2 - append second Via branch to Call-ID when sending command
to rtpproxy. See flag '1' for its meaning.
- + b - append branch specific variable to Call-ID when sending
- command to rtpproxy. See rtpproxy_offer() for details.
+ + b - append branch specific variable to Call-ID when sending
+ command to rtpproxy. See rtpproxy_offer() for details.
<listitem>
</listitem>
- t - do not include To tag to "delete" command to rtpproxy
- thus causing full call to be deleted. Useful for deleting
- unused rtpproxy call when 200 OK is received on a branch,
- where rtpproxy is not needed.
+ t - do not include To tag to "delete" command to rtpproxy thus
+ causing full call to be deleted. Useful for deleting unused
+ rtpproxy call when 200 OK is received on a branch, where
+ rtpproxy is not needed.
Example 1.16. rtpproxy_destroy usage
...
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 unforce_rtpproxy(), detecting
+ Manage the RTPProxy session - it combines the functionality of
+ rtpproxy_offer(), rtpproxy_answer() and unforce_rtpproxy(), detecting
internally based on message type and method which one to execute.
- It can take the same parameters as rtpproxy_offer(). The flags
- parameter to rtpproxy_manage() can be a configuration variable
+ It can take the same parameters as rtpproxy_offer(). The flags
+ parameter to rtpproxy_manage() can be a configuration variable
containing the flags as a string.
Functionality:
* If INVITE with SDP, then do rtpproxy_offer()
* If INVITE with SDP, when the tm module is loaded, mark transaction
- with internal flag FL_SDP_BODY to know that the 1xx and 2xx are
- for rtpproxy_answer()
+ with internal flag FL_SDP_BODY to know that the 1xx and 2xx are for
+ rtpproxy_answer()
* If ACK with SDP, then do rtpproxy_answer()
- * If BYE or CANCEL, or called within a FAILURE_ROUTE[], then do
+ * If BYE or CANCEL, or called within a FAILURE_ROUTE[], then do
unforce_rtpproxy()
* If reply to INVITE with code >= 300 do unforce_rtpproxy()
- * If reply with SDP to INVITE having code 1xx and 2xx, then do
- rtpproxy_answer() if the request had SDP or tm is not loaded,
+ * If reply with SDP to INVITE having code 1xx and 2xx, then do
+ rtpproxy_answer() if the request had SDP or tm is not loaded,
otherwise do rtpproxy_offer()
This function can be used from ANY_ROUTE.
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
+ Instruct the RTPproxy to stream prompt/announcement pre-encoded with
the makeann command from the RTPproxy distribution. The uac/uas suffix
- selects who will hear the announcement relatively to the current
+ selects who will hear the announcement relatively to the current
transaction - UAC or UAS. For example invoking the rtpproxy_stream2uac
- in the request processing block on ACK transaction will play the
- prompt to the UA that has generated original INVITE and ACK while
- rtpproxy_stop_stream2uas on 183 in reply processing block will play
- the prompt to the UA that has generated 183.
-
- Apart from generating announcements, another possible application of
- this function is implementing music on hold (MOH) functionality. When
- count is -1, the streaming will be in loop indefinitely until the
+ in the request processing block on ACK transaction will play the prompt
+ to the UA that has generated original INVITE and ACK while
+ rtpproxy_stop_stream2uas on 183 in reply processing block will play the
+ prompt to the UA that has generated 183.
+
+ Apart from generating announcements, another possible application of
+ this function is implementing music on hold (MOH) functionality. When
+ count is -1, the streaming will be in loop indefinitely until the
appropriate rtpproxy_stop_stream2xxx is issued.
- In order to work correctly, these functions require that a session in
+ In order to work correctly, these functions require that a session in
the RTPproxy already exists. Also those functions don't alter the SDP,
- so that they are not a substitute for calling rtpproxy_offer or
+ so that they are not a substitute for calling rtpproxy_offer or
rtpproxy_answer.
This function can be used from REQUEST_ROUTE, ONREPLY_ROUTE.
Meaning of the parameters is as follows:
- * prompt_name - name of the prompt to stream. Should be either
- absolute pathname or pathname relative to the directory where
+ * prompt_name - name of the prompt to stream. Should be either
+ absolute pathname or pathname relative to the directory where
RTPproxy runs.
- * count - number of times the prompt should be repeated. A value of
- -1 means that it will be streaming in a loop indefinitely, until
+ * count - number of times the prompt should be repeated. A value of
+ -1 means that it will be streaming in a loop indefinitely, until
the appropriate rtpproxy_stop_stream2xxx is issued.
Example 1.18. rtpproxy_stream2xxx usage
};
...
-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
+ Stop streaming of announcement/prompt/MOH started previously by the
+ respective rtpproxy_stream2xxx. The uac/uas suffix selects whose
announcement relatively to tha current transaction should be stopped -
UAC or UAS.
These functions can be used from REQUEST_ROUTE, ONREPLY_ROUTE.
-5.10. start_recording()
+5.10. start_recording()
- 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
+ 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.
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).
6.1. $rtpstat
Returns the RTP-Statistics from the RTP-Proxy. The RTP-Statistics from
- the RTP-Proxy are provided as a string and it does contain several
- packet-counters. The statistics must be retrieved before the session
- is deleted (before unforce_rtpproxy()).
+ the RTP-Proxy are provided as a string and it does contain several
+ packet-counters. The statistics must be retrieved before the session is
+ deleted (before unforce_rtpproxy()).
Example 1.20. $rtpstat-Usage
...
7.1. nh_enable_rtpp
- Enables a rtp proxy if parameter value is greater than 0. Disables it
+ Enables a rtp proxy if parameter value is greater than 0. Disables it
if a zero value is given.
- The first parameter is the rtp proxy url (exactly as defined in the
+ The first parameter is the rtp proxy url (exactly as defined in the
config file).
The second parameter value must be a number in decimal.
- NOTE: if a rtpproxy is defined multiple times (in the same or
- different sets), all of its instances will be enabled/disabled.
+ NOTE: if a rtpproxy is defined multiple times (in the same or different
+ sets), all of its instances will be enabled/disabled.
- Example 1.21. nh_enable_rtpp usage
+ Example 1.21. nh_enable_rtpp usage
...
$ kamctl fifo nh_enable_rtpp udp:192.168.2.133:8081 0
...
7.2. nh_show_rtpp
- Displays all the rtp proxies and their information: set and status
+ Displays all the rtp proxies and their information: set and status
(disabled or not, weight and recheck_ticks).
No parameter.
- Example 1.22. nh_show_rtpp usage
+ Example 1.22. nh_show_rtpp usage
...
$ kamctl fifo nh_show_rtpp
...
How can I report a bug?
Please follow the guidelines provided at:
- http://sip-router.org/tracker.
+ https://github.com/kamailio/kamailio/issues.
projects / sip-router / commitdiff