[Asterisk-code-review] res pjsip: Update endpoint transport option documentation. (asterisk[13])

[George Joseph]

George Joseph has submitted this change and it was merged. ( https://gerrit.asterisk.org/9537 )

Change subject: res_pjsip: Update endpoint transport option documentation.
......................................................................

res_pjsip: Update endpoint transport option documentation.

Change-Id: I5394fdff6a296efc8e1695a156e616acd932ae52
---
M configs/samples/pjsip.conf.sample
M res/res_pjsip.c
2 files changed, 18 insertions(+), 10 deletions(-)

Approvals:
  Kevin Harwell: Looks good to me, but someone else must approve
  George Joseph: Looks good to me, approved; Approved for Submit



diff --git a/configs/samples/pjsip.conf.sample b/configs/samples/pjsip.conf.sample
index f0dfdb5..e1ccd69 100644
--- a/configs/samples/pjsip.conf.sample
+++ b/configs/samples/pjsip.conf.sample
@@ -288,8 +288,8 @@
 ; a transport that you'll need to uncomment from the previous examples.
 ;
 ; Uncomment one of the transport lines to choose which transport you want. If
-; not specified then the default transport chosen is the first defined transport
-; in the configuration file.
+; not specified then the default transport chosen is the first compatible transport
+; in the configuration file for the contact URL.
 ;
 ; Modify the "max_contacts=" line to change how many unique registrations to allow.
 ;
@@ -668,7 +668,12 @@
 ;timers=yes     ; Session timers for SIP packets (default: "yes")
 ;timers_sess_expires=1800       ; Maximum session timer expiration period
                                 ; (default: "1800")
-;transport=     ; Desired transport configuration (default: "")
+;transport=     ; Explicit transport configuration to use (default: "")
+                ; This will force the endpoint to use the specified transport
+                ; configuration to send SIP messages.  You need to already know
+                ; what kind of transport (UDP/TCP/IPv4/etc) the endpoint device
+                ; will use.
+
 ;trust_id_inbound=no    ; Accept identification information received from this
                         ; endpoint (default: "no")
 ;trust_id_outbound=no   ; Send private identification details to the endpoint
diff --git a/res/res_pjsip.c b/res/res_pjsip.c
index e087564..0dbfc41 100644
--- a/res/res_pjsip.c
+++ b/res/res_pjsip.c
@@ -471,14 +471,17 @@
 					</para></description>
 				</configOption>
 				<configOption name="transport">
-					<synopsis>Desired transport configuration</synopsis>
-					<description><para>
-						This will set the desired transport configuration to send SIP data through.
+					<synopsis>Explicit transport configuration to use</synopsis>
+					<description>
+						<para>This will <emphasis>force</emphasis> the endpoint to use the
+						specified transport configuration to send SIP messages.  You need
+						to already know what kind of transport (UDP/TCP/IPv4/etc) the
+						endpoint device will use.
 						</para>
-						<warning><para>Not specifying a transport will <emphasis>DEFAULT</emphasis>
-						to the first configured transport in <filename>pjsip.conf</filename> which is
-						valid for the URI we are trying to contact.
-						</para></warning>
+						<note><para>Not specifying a transport will select the first
+						configured transport in <filename>pjsip.conf</filename> which is
+						compatible with the URI we are trying to contact.
+						</para></note>
 						<warning><para>Transport configuration is not affected by reloads. In order to
 						change transports, a full Asterisk restart is required</para></warning>
 					</description>

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Gerrit-Project: asterisk
Gerrit-Branch: 13
Gerrit-MessageType: merged
Gerrit-Change-Id: I5394fdff6a296efc8e1695a156e616acd932ae52
Gerrit-Change-Number: 9537
Gerrit-PatchSet: 1
Gerrit-Owner: Richard Mudgett <rmudgett at digium.com>
Gerrit-Reviewer: George Joseph <gjoseph at digium.com>
Gerrit-Reviewer: Jenkins2
Gerrit-Reviewer: Kevin Harwell <kharwell at digium.com>
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