[OpenSIPS-Devel] [ opensips-Feature Requests-3530685 ] b2bua unable to provide ringback

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Sun Jun 3 00:00:28 CEST 2012


Feature Requests item #3530685, was opened at 2012-05-29 18:15
Message generated for change (Comment added) made by rrevels
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Category: modules
Group: trunk
Status: Open
Priority: 5
Private: No
Submitted By: Richard Revels (rrevels)
Assigned to: Nobody/Anonymous (nobody)
Summary: b2bua unable to provide ringback

Initial Comment:
Using the prepaid example, when the media server sends the first BYE a re-INVITE is sent back to the caller from the b2bua with no SDP.  Next an invite is sent to the far end callee.  If the callee sends back a 183 w/ port information this is not able to be passed back to the caller as the caller is still waiting on a ACK (w/ sdp) to it's 200 to the re-INVITE.

If possible, I would rather see the b2bua send an ACK to the caller immediately with ip of 0.0.0.0 (call-on-hold).  Then re-INVITE the caller with the callee ports as soon as anything with an SDP replies.  Finally, if that something was a provisional response (or nothing with an sdp ever came through) then re-INVITE the caller again on receipt of the final 200 from the callee. 

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>Comment By: Richard Revels (rrevels)
Date: 2012-06-02 15:00

Message:
Yep.  This support is in place already.  Took a couple minutes to figure
out how to use the media url.  Here is a simple sample xml file with the
url being used if you want to add it to the b2bua tutorial.

<?xml version="1.0"?>
<scenario id="fax-detect" name="fax route" param="1" type="script">
	<init>
		<bridge>
			<server>
				<id>server1</id>
			</server>
			<client>
				<id>client1</id>
				<type>message</type>
				<destination>
					<value type="param">1</value>
				</destination>
			</client>
		</bridge>
		<state>1</state>
	</init>
	<rules>
		<request>
			<bye>
				<rule id="1">
					<condition>
						<state>1</state>
						<sender>
							<type>client</type>
							<id>client1</id>
						</sender>
					</condition>
					<action>
						<send_reply>
							<code>200</code>
							<reason>OK</reason>
						</send_reply>
						<delete_entity/>
						<bridge>
							<client>
								<id>server1</id>
							</client>
							<client>
								<id>client2</id>
								<destination>
									<value type="initial">server1</value>
								</destination>
							</client>
							<provisional_media>sip:ringback at sipserverthatanswerscall</provisional_media>
						</bridge>
						<state>2</state>
					</action>
				</rule>
			</bye>
		</request>
	</rules>
</scenario>

Call goes to media server which starts ringing then answers and does fax
detection for about 4 seconds.  If no fax detected then media server sends
BYE and call goes to to media server again which, this time, simply answers
and sends ring tones while the b2bua rings up the voice user for the
number.

Thank you Opensips Developers!  This is just such a versatile proxy and it
gets better every day.

This can be closed.  Use the Force, read the Source and all.

 


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Comment By: Richard Revels (rrevels)
Date: 2012-06-02 10:07

Message:
While looking through the release notes for opensips 1.6.4 (what can I say,
it's a slow day) I found an entry indicating this is already in place.  I'm
going to go back and look for problems in my configuration.

----------------------------------------------------------------------

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