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Unread 07-21-2007, 11:04 PM   #1
AlexErko
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Default VoXaLot Codecs

When one has chosen a VoIP service provider to include in a dialplan there is a provision for stipulating which codecs to use, or exclude, in relation to each of them, EXCEPT VoXaLot itself!

Some of my calls are VoXaLot to VoXaLot calls, and not being able to stipulate which codec is going to be used, is sometimes a major PITA, depending on the equipment and network being used.

Would someone please fix this oversight (?) ASAP.
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Unread 07-21-2007, 11:11 PM   #2
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Default VoXaLot to VoXaLot - RTP server codec settings not relevant

Quote:
Originally Posted by AlexErko View Post
When one has chosen a VoIP service provider to include in a dialplan there is a provision for stipulating which codecs to use, or exclude, in relation to each of them, EXCEPT VoXaLot itself!

Some of my calls are VoXaLot to VoXaLot calls, and not being able to stipulate which codec is going to be used, is sometimes a major PITA, depending on the equipment and network being used.
It's not relevant in the case of VoXaLot to VoXalot, as VoXaLot's RTP proxy is not used.

The codecs used will be whatever is negotiated between your SIP device and the endpoint SIP device where your SIP call is sent (ie, the codec settings on the SIP device of the VoXaLot user you are calling). The call will fail (eg with "488 Not Acceptable Here") if codec negotiation with the other device fails.

The best you can do is set the codecs you want to use in your SIP device, and hope that the other SIP device supports one of them.

Last edited by v164; 07-21-2007 at 11:14 PM.
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Unread 07-21-2007, 11:55 PM   #3
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Default

Quote:
Originally Posted by v164 View Post
It's not relevant in the case of VoXaLot to VoXalot, as VoXaLot's RTP proxy is not used.

The codecs used will be whatever is negotiated between your SIP device and the endpoint SIP device where your SIP call is sent (ie, the codec settings on the SIP device of the VoXaLot user you are calling). The call will fail (eg with "488 Not Acceptable Here") if codec negotiation with the other device fails.

The best you can do is set the codecs you want to use in your SIP device, and hope that the other SIP device supports one of them.
OK. I had misunderstood the situtation.
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Unread 07-21-2007, 11:55 PM   #4
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Quote:
Originally Posted by v164 View Post
It's not relevant in the case of VoXaLot to VoXalot, as VoXaLot's RTP proxy is not used.
Yes that's right. The only time Codecs come into play with Voxalot is when the call goes via one of your embedded providers.
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Unread 07-22-2007, 09:25 AM   #5
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Quote:
Originally Posted by martin View Post
Yes that's right. The only time Codecs come into play with Voxalot is when the call goes via one of your embedded providers.
.
Hi,martin

Do all Sip Broker access gateways support the same CODECS?

Assume my ATA registered at Voxalot, only supports G.711-A, a Sip Broker access gateway only supports G.711-U (not support G.711-A). A caller access this Sip Broker number and dial my Voxalot account " *061-25XXXX ". The call will fail? or the transcoding will happen?

Thanks.
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Unread 07-22-2007, 09:28 AM   #6
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Originally Posted by hust View Post
Hi,martin

Do all Sip Broker access gateways support the same CODECS?

Assume my ATA registered at Voxalot, only supports G.711-A, a Sip Broker access gateway only supports G.711-U (not support G.711-A). A caller access this Sip Broker number and dial my Voxalot account " *061-25XXXX ". The call will fail? or the transcoding will happen?

Thanks.
Transcoding will happen.
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Unread 07-22-2007, 09:56 AM   #7
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Quote:
Originally Posted by martin View Post
Transcoding will happen.
.
Thanks.

SIP broker is a SIP broker.
Voxalot is a VSP broker for us.
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Unread 07-24-2007, 02:41 PM   #8
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Default transcoding on incoming SIP calls

Quote:
Originally Posted by martin View Post
Transcoding will happen.
Are you sure about that, or is this a special arrangement for the SIP Broker PSTN access numbers?


I've got my Atcom AT-530 SIP phone registered to us.voxalot.com. It does not support the gsm codec.

If I send a SIP call to *010 <voxalot number> @sipbroker.com using the X-Ten Lite softphone, stipulating the gsm codec only, the call fails with "480 Temporarily Unavailable":

SEND >> 64.34.162.221:5060
Code:
INVITE sip:*01066xxxx@sipbroker.com SIP/2.0
Via: SIP/2.0/UDP 221.xx.xx.yyy:5060;branch=z9hG4bK9FD9C80863834DB7BC500FD5D6F03974
From: 8150xxxxxxxx <sip:8150xxxxxxxx@sipbroker.com>;tag=2795761264
To: <sip:*01066xxxx@sipbroker.com>
Contact: <sip:8150xxxxxxxx@221.xx.xx.yyy:5060>
Call-ID: 7101315C-4947-47C6-BF67-CE135679335F@221.xx.xx.yyy
CSeq: 55933 INVITE
Max-Forwards: 70
Content-Type: application/sdp
User-Agent: X-Lite release 1103m
Content-Length: 201

v=0
o=8150xxxxxxxx 1782021 1782116 IN IP4 221.xx.xx.yyy
s=X-Lite
c=IN IP4 221.xx.xx.yyy
t=0 0
m=audio 29360 RTP/AVP 3 101
a=rtpmap:3 gsm/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
...

RECEIVE << 64.34.162.221:5060
Code:
SIP/2.0 480 Temporarily Unavailable
Via: SIP/2.0/UDP 221.xx.xx.yyy:5060;branch=z9hG4bK9FD9C80863834DB7BC500FD5D6F03974
From: 8150xxxxxxxx <sip:8150xxxxxxxx@sipbroker.com>;tag=2795761264
To: <sip:*01066xxxx@sipbroker.com>;tag=54333a515cb8c1bd95fa9f9db754f4d6-ffdd
Call-ID: 7101315C-4947-47C6-BF67-CE135679335F@221.xx.xx.yyy
CSeq: 55933 INVITE
Server: OpenSer (1.1.0-notls (i386/linux))
Content-Length: 0
Warning: 392 64.34.163.35:5060 "Noisy feedback tells:  pid=30938 req_src_ip=64.34.162.221 req_src_port=5060 in_uri=sip:66xxxx@proxy01.us1.voxalot.com out_uri=sip:66xxxx@proxy01.us1.voxalot.com via_cnt==3"

The call is accepted if I select the ulaw codec:

Code:
SEND >> 64.34.162.221:5060
INVITE sip:*01066xxxx@sipbroker.com SIP/2.0
Via: SIP/2.0/UDP 221.xx.xx.yyy:5060;branch=z9hG4bK3AB3665BC6344878A418C081E866D048
From: 8150xxxxxxxx <sip:8150xxxxxxxx@sipbroker.com>;tag=3531632047
To: <sip:*01066xxxx@sipbroker.com>
Contact: <sip:8150xxxxxxxx@221.xx.xx.yyy:5060>
Call-ID: 3A85AB83-874C-412F-B03C-CF0C26560A4D@221.xx.xx.yyy
CSeq: 29057 INVITE
Max-Forwards: 70
Content-Type: application/sdp
User-Agent: X-Lite release 1103m
Content-Length: 202

v=0
o=8150xxxxxxxx 1769675 1769766 IN IP4 221.xx.xx.yyy
s=X-Lite
c=IN IP4 221.xx.xx.yyy
t=0 0
m=audio 29360 RTP/AVP 0 101
a=rtpmap:0 pcmu/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
...

RECEIVE << 64.34.162.221:5060
Code:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 221.xx.xx.yyy:5060;branch=z9hG4bK3AB3665BC6344878A418C081E866D048
Record-Route: <sip:64.34.173.199;lr=on;ftag=3531632047>
Record-Route: <sip:64.34.163.35;lr=on;ftag=3531632047>
Record-Route: <sip:64.34.162.221;lr=on;ftag=3531632047>
From: 8150xxxxxxxx <sip:8150xxxxxxxx@sipbroker.com>;tag=3531632047
To: <sip:*01066xxxx@sipbroker.com>;tag=22037778
Call-ID: 3A85AB83-874C-412F-B03C-CF0C26560A4D@221.xx.xx.yyy
CSeq: 29057 INVITE
Contact: <sip:*01066xxxx@221.xx.xx.xx:6080>
supported: replaces
Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, PRACK, UPDATE, MESSAGE
Content-Type: application/sdp
Content-Length: 151

v=0
o=sdp_admin 21487122 24659200 IN IP4 221.xx.xx.xx
s=A conversation
c=IN IP4 221.xx.xx.xx
t=0 0
m=audio 10016 RTP/AVP 0
a=rtpmap:0 PCMU/8000
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Unread 07-24-2007, 03:58 PM   #9
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Quote:
Originally Posted by v164 View Post
Are you sure about that, or is this a special arrangement for the SIP Broker PSTN access numbers?
I think it is a special arrangement only for the SIP Broker PSTN access numbers & Voxalot(*010 *031 *061) users. otherwise, Voxalot RTP proxy server will ....

Another : assume my ATA registered at Voxalot, only supports G.723, i use ATA to dial *747-1-747XXXXXXX(a Gizmoproject num) via Sipbroker, as i know, Gizmo does not support G.723. Voxalot will not transcode and the call will fail.

Maybe, Voxalot should make a form to show when transcoding will happen and when the call will fail for CODECS match and all Voxalot supporting CODECS...all Voxalot supporting transcoding types . It also can help members improve voice quality by ourselves.

Last edited by hust; 07-24-2007 at 05:13 PM.
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