Forcing a codec

syntax1269 · Brekeke Member Joined: 18 Nov 2007 Posts: 16

1. Brekeke Product Name and version:
Brekeke PBX, Version 2.0.7.2 Pro Evaluation
ID: 00000000 , Users: 10 , Concurrent: 10

2. Java version:
1.5 update 14
3. OS type and the version:
Windows 2003 server with SP2

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

5. Select your network pattern from http://www.brekeke-sip.com/bbs/network/networkpatterns.html : Pattern 9

6. Your problem:
I am tring to force a codec (0, 98) any one that the trial will support. i have tried adding the setting
&net.rtp.audio.payloadtype=0
into BSS dial plan, but for some reason it is still picking 18 and not 0.

i have been at this for a few days now, reading the forum and i am still stuck. i can register no problem. the PBX answers the call, and "tries" to play the auto attendent message but i get back only mis-matched codec sounds.

voipwell.com · Posted: Mon Nov 19, 2007 9:45 pm Post subject:

What dial plan did you put it in. Also there is a question in the sip config and pbx config that relates to it you want the codec to be negotiated or forced to use what you put into the pbx config screen.

syntax1269 · Brekeke Member Joined: 18 Nov 2007 Posts: 16

voipwell.com · Posted: Tue Nov 20, 2007 8:23 am Post subject:

Hi,

Have you verified that Brekeke is using this dial plan by looking at active session detail during a call?

syntax1269 · Brekeke Member Joined: 18 Nov 2007 Posts: 16

When calling out it is stuck at Provisioning and the call never goes through.
Session Details
From-uri sip:17772377361@callcentric.com:5060
From-ip 209.26.247.86:15062
From-if 209.26.247.86:5060
To-uri sip:17771234567@callcentric.com
To-ip 204.11.192.23
To-if 209.26.247.86:5060
Call-ID 22225377-7c57dd94-459b46f8-e9de444a@209.26.247.86
rule Callcentric
plug-in InviteSession
sip-packet-total 5
listen-port 5060
sip-packet-stacked 0
session-status Provisioning
time-inviting Tue Nov 20 10:55:29 EST 2007
rtp-relay on
rtp-srcdst
rtp-dstsrc
media audio
transport RTP/AVP
payload -
status active
listen-port 10052
send-port
target 209.26.247.86:11058
packet-count 0
packet/sec 0
current size 0
buffer size 260

when calling into the SIP server
Session Details
From-uri sip:13527293429@callcentric.com
From-ip 204.11.192.23:5060
From-if 209.26.247.86:5060
To-uri sip:17772377361@209.26.247.86:5060
To-ip 127.0.0.1:15060
To-if 209.26.247.86:5060
Call-ID 14213_1195334004@192.168.1.30
rule To PBX
plug-in InviteSession
sip-packet-total 6
listen-port 5060
sip-packet-stacked 0
session-status Accepted
time-inviting Tue Nov 20 11:00:00 EST 2007
rtp-relay on
rtp-srcdst
media audio
transport RTP/AVP
payload 18 (G729/8000)
status active
listen-port 10056
send-port 10054
target 209.26.247.86:11060
packet-count 170
packet/sec 56
current size 32
buffer size 260
rtp-dstsrc
media audio
transport RTP/AVP
payload 0 (PCMU/8000)
status active
listen-port 10054
send-port 10056
target 204.11.192.23:51742
packet-count 1
packet/sec 0
current size 172
buffer size 260

voipwell.com · Posted: Tue Nov 20, 2007 7:04 pm Post subject:

Hi,

Ok, now I remember the Callcentric details. I had tried to get Callcentric to work last spring and wasn't able to. Another poster did finally figure out what it was. First, you need this in the deploy dial plan. &net.sip.via.multple=false

Then the poster found out that Callcentric g711 uses 40ms packets which Brekeke doesn't support so he had to turn off relay. So, turn off rtp relay for this ARS in the ARS tab. Then review this current rtp relay disabling thread to see how to turn of rtp relay for these calls.

http://www.brekeke-sip.com/bbs/viewtopic.php?t=5313

That should do it for you.

syntax1269 · Brekeke Member Joined: 18 Nov 2007 Posts: 16

this is the responce from Callcentric

------START--------
Nov 20, 2007 04:08 PM | Customer service
Hello,

I have just contacted one of our engineers on this particular PBX and it seems that the Brekeke PBX does not properly work with our services. We have been able to get it to register in one circumstance however there is still the unexplainable, on our side, discrepancies in Ondo's SIP implementation which prevents their users from properly using Callcentric's services.

We apologize but our recommendation is to either use a different PBX solution, such as Asterisk./freePBX/trixbox (http://www.callcentric.com/support/), or another commercial vendor who properly follows the SIP standard. That is until Ondo decides to fix the issues with their software.
------END------

so i guess Brekeke needs to fix this. :-(
unless someone else has any sugestions for me to try.

syntax1269 · Brekeke Member Joined: 18 Nov 2007 Posts: 16

voipwell,

yeah i have been reading your posts Smile

i will try that last thing and i will post my findings.

hopefully this will work and i can start to build this PBX/SIP Server out.

thank you very much!

voipwell.com · Posted: Tue Nov 20, 2007 7:15 pm Post subject:

hmmmmm...

So Callcentric forces 40ms G711u packets, and doesn't allow you to use another size packet and then claims Brekeke is wrong because they don't allow you to use another size packet other than 30ms. So when Callcentric uses a larger packet more prone to internet problems and can't change it they are right and when Brekeke uses a smaller packet less likely to fragment over the internet Callcentric claims you should dump them for a package that does what Callcentric can't do.

All I can say is if you want to use a provider with 40ms packets it will never be a quality voice stream and most certainly never meet business quality standards. I don't blame Brekeke for not allowing the g711u packet size to blow up to 40ms. It insures that their users will enjoy better voice quality with smaller voice packets. We are a vsp and we generously give all our users 20ms packets resulting in the best call quality available. Callcentric uses 40ms packets because it saves some bandwith and processing power. 40ms packets need less overhead and it cuts their packets per second in half. That's acceptable if they are flexible and allow you to use a smaller packet if you wish but they have locked it in at 40ms which deprives you of the packet quality you can get from just about every other vsp out there.

syntax1269 · Brekeke Member Joined: 18 Nov 2007 Posts: 16

that is exactly what i was thinking for g711
but i do have them on an altigen PBX, and using G729 and the quality is perfect. some of my users say it is better, but who knows.

maybe there is a different standard/restriction on g729 as far as packets being 20ms or 30ms.

but this is just testing now, and don't want to spend the $$$ for the server and then buy g729 licenses jsut to find out somehting works or worse yet doesn't work. and i will then have to go with another provider.

unless you know of some other provider that has the same or less price as callcentric.

voipwell.com · Posted: Wed Nov 21, 2007 8:15 am Post subject:

Hi,

You don't need g729 licenses to use g729 with brekeke. You only would need g729 license to use voicemail, conference and other pbx features. You must specify in the ARS that rtp relay is off. Rtp relay in pbx is different than in sip server. RTP relay in pbx relates to relaying the rtp stream to brekeke's media server software and RTP relay in sip server relates to relaying rtp stream from phone to vsp. You must have sip server rtp relay on to get the stream from your vsp to the phones behind nat, but you can have rtp relay off in the pbx to prevent the stream going thru the media server which requires g729 license. I think the standard for g729 is 20ms but with your vsp who knows. At least you would be able to test it.

What about Callcentric's pricing is attracting you? Is it the domestic rate or overseas rate?

syntax1269 · Brekeke Member Joined: 18 Nov 2007 Posts: 16

voipwell,

maybe i am setting something wrong then.
this is the current config..

in BSS:
Callcentric rule:
Matching Patern:
$request=^INVITE
From=sip:(.+)@
To=callcentric\.com
$port=15062

Deploy:
Contact=<sip:%1@callcentric.com:5060>
&net.sip.via.multple=0
$rtp=false
&net.rtp.relay.localhost=off

-------
PBX ARS

Callcentric:
Patterns - OUT
RTP relay = off
Codec Priority = 98,0
Block SIP INFO (DTMF) = no
Send RTCP = off
Session Timer(sec, 0=disable) = 0
100rel = off
Next route on dailure = no
Disable on registration failure = no
Response timeout (ms) = -1
Error codes = 500-599
Recovery time (ms) = 0
Disable all OUT patterns on failure = yes

as far as why Callcentric? domestic LD is 0.0198 is one reason coming from Embarq PRI LD/Local .. the second reson is that it works with our Altigen PBX (very fussy when it comes to VSP's) and with that service, using ofcourse g729 at 20ms, we have had no issues. so i figured why not try them with other SIP servers.