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/ ... terns.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
RTP-Relay
Moderator: Brekeke Support Team
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
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