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RTP-Relay
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dahleh
Brekeke Junior Member


Joined: 18 Feb 2007
Posts: 9

PostPosted: Thu Oct 11, 2007 10:32 am    Post subject: RTP-Relay Reply with quote

1. Brekeke Product Name and version:2.0.7.2

2. Java version:1.5.0_09

3. OS type and the version:WINXP

4. UA (phone), gateway or other hardware/software involved:

5. Select your network pattern from http://www.brekeke-sip.com/bbs/network/networkpatterns.html :

6. Your problem:

How can I turn off RTP-Relay for incoming calls, I have turned it off for outgoing calls with this dial plan

Matching Patterns
$request = ^INVITE
$target = hostserver
Deploy Patterns
From = sip:(1.+)@xxx\.xxx\.xxx\.xxx\:
$rtp = false

I can make calls out with RTP-Relay OFF but all calls coming in have RTP-Relay on
All Iam doing is forwarding all outgoing and incoming calls to my service provider thorugh a different port than 5060

I aprreciate any input. Thx
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jelly
Brekeke Talented


Joined: 20 Jun 2007
Posts: 62

PostPosted: Thu Oct 11, 2007 11:28 am    Post subject: Reply with quote

are you using Brekeke PBX? if so, go to pbx admintool > options > setting > pbx system settings and set rtp = off
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dahleh
Brekeke Junior Member


Joined: 18 Feb 2007
Posts: 9

PostPosted: Thu Oct 11, 2007 12:53 pm    Post subject: Reply with quote

Thank you for your reply jelly

I am using SIP Server not the PBX and I have there are only two settings in the RTP and I have them as follows:

RTP Relay = auto
RTP Relay (UA on this machine) = off

The first one has only the auto or on option

Thx
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dahleh
Brekeke Junior Member


Joined: 18 Feb 2007
Posts: 9

PostPosted: Thu Oct 11, 2007 1:02 pm    Post subject: Reply with quote

by the way these are the details of one incoming call that has the RTP turned on

From-uri sip:+1phone#@hostserver
From-ip hostserver:5060
From-if sipserverIP:port
To-uri sip:1phone#@sipserverip:port [behind NAT]
To-ip callerpublicIPaddress:port
To-if sipserverIP:port
Call-ID NYCMGC0120071011184530000897@209.244.63.25
rule registered=sip:phone#(sip:phone#@192.168.1.2:7777)
plug-in InviteSession
sip-packet-total 10
listen-port port
sip-packet-stacked 0
session-status Closing
time-inviting Thu Oct 11 12:46:03 MDT 2007
time-talking Thu Oct 11 12:46:13 MDT 2007
length-talking 01:07:58
rtp-relay on
rtp-srcdst
media audio
transport RTP/AVP
payload 18 (G729/8000)
status active
listen-port 10746
send-port 10744
target phone#:16466
packet-count 134829
packet/sec 32
current size 42
buffer size 260
rtp-dstsrc
media audio
transport RTP/AVP
payload 18 (G729/8000)
status active
listen-port 10744
send-port 10746
target serviceprovider IP:45866
packet-count 135435
packet/sec 32
current size 42
buffer size 260
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pwestly
Brekeke Newbie


Joined: 25 Sep 2007
Posts: 4

PostPosted: Thu Oct 11, 2007 11:26 pm    Post subject: Reply with quote

you can try this

Matching Patterns
$registered = true
Deploy Patterns
$rtp = false

for all registered ua, rtp relay will be turned off.
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dahleh
Brekeke Junior Member


Joined: 18 Feb 2007
Posts: 9

PostPosted: Fri Oct 12, 2007 7:56 am    Post subject: Reply with quote

Thank you so much, it worked ..
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