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Unread 03-24-2007, 02:37 PM   #1
v164
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Default Can't get voice packets to bypass VoXaLot

As Martin has confirmed in this thread,

"Which VoXaLot services are B2BUA?"
http://forum.voxalot.com/showthread.php?t=1258

when making SIP calls through VoXaLot, the RTP voice packets for your call can (will?) bypass VoXaLot if your SIP device is correctly configured (ie, use of STUN, etc if behind NAT, and codec settings so that VoXaLot doesn't need to transcode).

Having your voice packets bypass VoXaLot is ideal, because it means you get all the benefits of VoXaLot routing your calls (dial plans, ENUM lookups, speed dial, SIP-code dialing), with no loss of voice quality (ie, no extra latency), so it is worthwhile putting in the effort to ensure your SIP device is setup optimally.

For example, in my situation:

My SIP phone is in Japan,

VoXaLot's SIP server (us.voxalot.com) is in the United States, and

my provider's SIP server (sip.pennytel.com) is in Sydney (Australia).


The SIP INVITE message will transit us.voxalot.com (where the dialed number will be checked against ENUM and dial plans) on the way to sip.pennytel.com, however, ideally, the voice packets will travel direct between my SIP phone and sip.pennytel.com.


However, I regret to say that I have been unable to get us.voxalot.com to stop proxying the voice packets.

My SIP phone is sitting on my public IP address, and to make sure VoXaLot doesn't do "NAT assistance", I have set "Enable symmetric NAT handling" to "No" on the member details page:


"Enable symmetric NAT handling (if unsure, set to "Yes")"
http://www.voxalot.com/action/memberDetails


That leaves the issue of codec settings, codec negotiation, and transcoding.

What I'm seeing is that us.voxalot.com is proxying the voice packets even when both ends of the call are using the same codec, and so far I haven't been able to prevent this.


Basically, I just want to use the G711 ulaw codec with Pennytel. In the Provider Details page, I've entered ulaw as the only codec. In my SIP phone, I can't disable the other codecs, but I can put ulaw at the top of the list.

As an example, for a call to an Australian 1800 number, here is the SIP INVITE message my SIP phone sends to us.voxalot.com:
Code:
INVITE sip:61180050xxxx@us.voxalot.com SIP/2.0
Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:6080;branch=z9hG4bK219075768430918035;rport
From: 660xxx <sip:660xxx@us.voxalot.com>;tag=674623933
To: "61180050xxxx" <sip:61180050xxx@us.voxalot.com>
Call-ID: 2272424319-451924324394@xxx.xxx.xxx.xxx
CSeq: 1 INVITE
Contact: <sip:660xxx@xxx.xxx.xxx.xxx:6080>
max-forwards: 70
supported: 100rel
user-agent: Voip Phone 1.0
Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, SUBSCRIBE, PRACK, UPDATE
Content-Type: application/sdp
Content-Length: 342  

v=0
o=sdp_admin 68861768 17995204 IN IP4 xxx.xxx.xxx.xxx
s=A conversation
c=IN IP4 xxx.xxx.xxx.xxx
t=0 0
m=audio 10070 RTP/AVP 0 4 4 18 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:4 G723/8000
a=rtpmap:4 G723high/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv

us.voxalot.com then sends a SIP INVITE message like this to sip.pennytel.com:
Code:
INVITE sip:61180050xxxx@sip.pennytel.com:5060 SIP/2.0
Via: SIP/2.0/UDP 64.34.173.199:5061;branch=z9hG4bK44e1d1a8;rport
From: "660xxx" <sip:8886xxxxxx@sip.pennytel.com>;tag=as1ee2c8bc
To: <sip:61180050xxxx@sip.pennytel.com:5060>
Contact: <sip:8886xxxxxx@64.34.173.199:5061>
Call-ID: 0b923187704ffc7e0b5aaa507bacbfca@sip.pennytel.com
CSeq: 102 INVITE
User-Agent: VoXaLot
Max-Forwards: 70
Date: Sat, 24 Mar 2007 10:52:24 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 218

v=0
o=root 18772 18772 IN IP4 64.34.173.199
s=session
c=IN IP4 64.34.173.199
t=0 0
m=audio 15906 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
The SDP body shows that VoXaLot is asking sip.pennytel.com to send the RTP packets to 64.34.173.199 (ie, us.voxalot.com).

I'm not sure if I properly understand the codec negotiation mechanism when calls are routed through VoXaLot. I would have thought that, provided all of the codecs entered in the "Provider Details" page are offered by the SIP phone in it's SDP body, then VoXaLot would just forward that SDP body on to the provider as is. However, I'm starting to think that maybe VoXaLot starts by always proxying the media (to help ensure that the call can be established reliably), and then relies on the provider to initiate or correctly respond to a SIP "re-INVITE" message to subsequently modify the media stream. If this is the case, then it means that VoXaLot could continue to proxy the voice packets, even when no transcoding is necessary.

Martin's quote made me start thinking this.

Quote:
"...or because the destination UAC is not initiating a re-invite."
Does that mean that having voice packets bypass VoXaLot depends on sip.pennytel.com initiating or supporting a re-INVITE ? And if sip.pennytel.com doesn't support that, then there is no way to get the voice packets to bypass VoXaLot?
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