Commit Graph

5 Commits

Author SHA1 Message Date
Joshua Colp
3c36c29c81 res_hep_rtcp: Provide chan_sip Call-ID for RTCP messages.
This change adds the required logic to allow the SIP
Call-ID to be placed into the HEP RTCP traffic if the
chan_sip module is used. In cases where the option is
enabled but the channel is not either SIP or PJSIP then
the code will fallback to the channel name as done
previously.

Based on the change on Nir's branch at:
team/nirs/hep-chan-sip-support

ASTERISK-26427

Change-Id: I09ffa5f6e2fdfd99ee999650ba4e0a7aad6dc40d
2017-05-09 05:38:59 -05:00
Matt Jordan
05713c36ea configs/samples/hep.conf.sample: Clarify how the HEP stack works
This patch updates the documenation in hep.conf.sample to better specify
how the various HEP modules interact.

ASTERISK-26717 #close

Change-Id: I337fb742a89e3ec5edc7fc7a7a0295218d841124
2017-03-14 09:52:59 -06:00
Matt Jordan
dab2a6b689 hep.conf.sample: Default 'enabled' to 'no'
Following the principle of least surprise, we should not be sending
massive numbers of PJSIP and RTCP HEP packets out into the ether to some
only-slightly-random IP address. Having 'enabled' set to 'no' in the
sample configuration file should prevent this from happening for those
who run 'make samples'.

ASTERISK-26159 #close

Change-Id: I1753a64ca83a3442a6ebdc31061f8185c062d9b1
2016-06-29 16:18:53 -05:00
Matt Jordan
e06a23681c res_hep: Provide an option to pick the UUID type
At one point in time, it seemed like a good idea to use the Asterisk
channel name as the HEP correlation UUID. In particular, it felt like
this would be a useful identifier to tie PJSIP messages and RTCP
messages together, along with whatever other data we may eventually send
to Homer. This also had the benefit of keeping the correlation UUID
channel technology agnostic.

In practice, it isn't as useful as hoped, for two reasons:
1) The first INVITE request received doesn't have a channel. As a
   result, there is always an 'odd message out', leading it to be
   potentially uncorrelated in Homer.
2) Other systems sending capture packets (Kamailio) use the SIP Call-ID.
   This causes RTCP information to be uncorrelated to the SIP message
   traffic seen by those capture nodes.

In order to support both (in case someone is trying to use res_hep_rtcp
with a non-PJSIP channel), this patch adds a new option, uuid_type, with
two valid values - 'call-id' and 'channel'. The uuid_type option is used
by a module to determine the preferred UUID type. When available, that
source of a correlation UUID is used; when not, the more readily available
source is used.

For res_hep_pjsip:
 - uuid_type = call-id: the module uses the SIP Call-ID header value
 - uuid_type = channel: the module uses the channel name if available,
                        falling back to SIP Call-ID if not
For res_hep_rtcp:
 - uuid_type = call-id: the module uses the SIP Call-ID header if the
                        channel type is PJSIP and we have a channel,
                        falling back to the Stasis event provided
                        channel name if not
 - uuid_type = channel: the module uses the channel name

ASTERISK-25352 #close

Change-Id: Ide67e59a52d9c806e3cc0a797ea1a4b88a00122c
2016-05-14 09:42:20 -05:00
Matthew Jordan
fc0fecb476 configs: Move sample config files into a subdirectory of configs
This moves all samples configs from configs/ to configs/samples. This allows
for additional sets of sample configuration files to be added in the future.

Review: https://reviewboard.asterisk.org/r/3804/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418870 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-07-17 21:17:28 +00:00