No voice with G729 through BSS

1. Brekeke Product Name and version: BSS 2.2.7.8

2. Java version: 10.4.2.1

3. OS type and the version: Windows 2003 Enterprise

4. UA (phone), gateway or other hardware/software involved: SVK Software IVRCAt, Radiuscat

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

6. Your problem: I need to force G.729 codec through BSS for outbound calls. because my ITSP doesn't support G.711. Also i am unbale to use Brekeke PBX to force G.729 because ITSP only works with IP secenario(can't register with user name ans password). Here is my BSS secession after it connected to the destination and i COULDNOT hear the destination but he was able to hear me fine. I used dial plan &net.rtp.audio.payloadtype = 18 in the BSS deploy pattern. Can you please tell me how to fix this?

Active Sessions > DetailsSession ID : 15

Session Details
EX-SID 14
From-uri sip:2234@192.168.2.13:5060
From-ip 192.168.2.13:55062 (UDP)
From-if 192.168.2.13:5060
To-uri sip:14162982400@77.72.169.128
To-ip 75.79.19.25 (UDP)
To-if 65.125.235.225:5060
Call-ID b043d4bd-72404674-2701f474-aae07969@192.168.1.13
rule Termination Siptraff
plug-in InviteSession
sip-packet-total 4
listen-port 5060
session-status Talking
time-inviting Wed Jun 08 12:38:52 EDT 2011
time-talking Wed Jun 08 12:38:55 EDT 2011
length-talking 00:00:07
rtp-relay on
rtp-srcdst
media audio
transport RTP/AVP
payload 0 (PCMU/8000)
status active
listen-port 10006
send-port 10004
target 65.45.83.45:8752
packet-count 363
packet/sec 36
current size 172
buffer size 260
rtp-dstsrc
media audio
transport RTP/AVP
payload 18 (G729/8000)
status active
listen-port 10004
send-port 10006
target 192.168.2.13:31210
packet-count 356
packet/sec 35
current size 32
buffer size 260

Thanks

is your bss behind router?
check wiki page below for portforwarding and set router global ip at bss/configuration/system/network/interface address

http://wiki.brekeke.com/wiki/Set-port-forwarding

Greatly appreciate your answer Hope! As you mentioned i forwarded the ports and my BSS already had the global IP address in there but still no luck.. Same issue. I also wonder the " payload 0 (PCMU/8000) " which looks like still using the G.711 codec on my secession details above..what do you think?

can caller accept g729?
if not, need to set g729 at caller side.
Brekeke sip server doesnot handle conversion between different codecs, brekeke pbx can convert codecs.

Thanks for reply Hope! Im using IVR Cat (Based on JTAPI from SVK Software) as an IVR for my calling card which doesn't support G.729. So when you mentioned the conversion you mean the IVR supposed the do the conversion when the packets received by BSS?
Why is the secession details shows two different "payload" lines which one shows G.729 and other line shows PCMU?

Thanks for your help again.

Ajanth

if client didnot support the codec, it should return error message, and call cannot be established, i'm not sure why IVR cat can accept it.

Thanks for your continous support Hope! Here is a liitle more explanation, when i use Brekeke PBX with G.729 as priority codec it works fine, but when i use the BSS with dilaplan &net.rtp.audio.payloadtype = 18 no voice both sides. But when i use BSS with the dilaplan &net.rtp.audio.payloadtype = 4 ,it works fine. And i am sure it will not be the vendor because it works fine through Brekeke PBX. You have any idea?

brekeke pbx can convert codecs between clients,
the call through brekeke pbx has two sessions:
one is caller to brekeke pbx, the other is Brekeke pbx to callee.
If one client can only accept codec g729 and the other only accept codec G711u, Brekeke pbx can use g729 codec for the call with one client and use G711u for the call with the other client when Brekeke pbx has G729 codec setting. but there is no such feature in brekeke sip server, when set &net.rtp.audio.payloadtype in the dial plan, both clients will use codec set in &net.rtp.audio.payloadtype.

Thanks Hope! That explains what cause the voice problem because the IVRCat only support G711 and iLBc and it can't do the conversion. so I am stuck here Do you know any other Pre IVR for callingcards that supports Brekeke?

Thanks
Ajanth