About Codec
Moderator: Brekeke Support Team
-
- Posts: 23
- Joined: Mon Mar 07, 2011 7:48 am
About Codec
1. Brekeke Product Name and Version:
Version 3.2.4.3 Standard
Version 2.4.7.3 Standard
2. Java version:6
3. OS type and the version:win2003
4. UA (phone), gateway or other hardware/software involved:yealink
5. Your problem:
I have two sip servers, Server A's version is 2.4.7.3,Server B's version is 3.2.4.3.
I add the dailplan in Server A:
Matching Patterns:
$request=^INVITE
To=sip:(1[0-9]{10})@
$geturi(From)=(.+)
Deploy Patterns:
From="%1"<%2>
To=sip:%1@192.168.66.30:5060
All phones register to Server A.And 192.168.66.30 is server B's ip address.
And I add the dailplan in Server B:
Matching Patterns:
$request = ^INVITE
To = sip:(139[0-9]{8})@
$geturi(From) = (.+)
Deploy Patterns:
$auth = false
&net.rtp.audio.payloadtype = 18
From = "%1"<%2>
To = sip:%1@192.168.2.10:2080
192.168.2.10 is a gateway of E1's ip address.And Server B' B2B-UA mode is on.
Most of time the call can reach the gateway and dail out.But about 10% calls can not call out.At this time,I view the Active Sessions and see the session's detail,I found under these circumstances the payload's values is empty,and in the log,the failure call's error code is 500 or 617 or 603.How do I change the setting?
Thanks you!
Version 3.2.4.3 Standard
Version 2.4.7.3 Standard
2. Java version:6
3. OS type and the version:win2003
4. UA (phone), gateway or other hardware/software involved:yealink
5. Your problem:
I have two sip servers, Server A's version is 2.4.7.3,Server B's version is 3.2.4.3.
I add the dailplan in Server A:
Matching Patterns:
$request=^INVITE
To=sip:(1[0-9]{10})@
$geturi(From)=(.+)
Deploy Patterns:
From="%1"<%2>
To=sip:%1@192.168.66.30:5060
All phones register to Server A.And 192.168.66.30 is server B's ip address.
And I add the dailplan in Server B:
Matching Patterns:
$request = ^INVITE
To = sip:(139[0-9]{8})@
$geturi(From) = (.+)
Deploy Patterns:
$auth = false
&net.rtp.audio.payloadtype = 18
From = "%1"<%2>
To = sip:%1@192.168.2.10:2080
192.168.2.10 is a gateway of E1's ip address.And Server B' B2B-UA mode is on.
Most of time the call can reach the gateway and dail out.But about 10% calls can not call out.At this time,I view the Active Sessions and see the session's detail,I found under these circumstances the payload's values is empty,and in the log,the failure call's error code is 500 or 617 or 603.How do I change the setting?
Thanks you!
-
- Posts: 23
- Joined: Mon Mar 07, 2011 7:48 am
Thank for your reply.janP wrote:> &net.rtp.audio.payloadtype = 18
Do you know what it means?
The payload type 18 means G729.
Both UA support G729 audio codec?
I didn't add &net.rtp.audio.payloadtype = 18 at first,But when 617 error appears,I try to add it.
Both UA support G729,if not,all calls will fail,But now only 10% calls fail,and I have checked the concurrent number of G729 of the gateway,It's enough.
-
- Posts: 23
- Joined: Mon Mar 07, 2011 7:48 am
-
- Posts: 23
- Joined: Mon Mar 07, 2011 7:48 am
-
- Posts: 23
- Joined: Mon Mar 07, 2011 7:48 am