My SIP Switch with your provider

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TheFug
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Joined: Sat Oct 06, 2007 8:23 am
Location: The Netherlands, North-Holland

Post by TheFug » Thu Jan 21, 2010 8:59 pm

You can register the account on a ATA or softphone ?
You can also add :5060 behind the sip server uri/address, or when using a sip-uri.
You can't do much more than this, in normal situations you could opt for STUN or proxy servers.
Thanks, The Fug.

gear: my ISP's Zyxel Modem/Router in bridge, Sitecom WL309 Router, Siemens Gigaset 301D

philoum78
Posts: 23
Joined: Mon Jul 06, 2009 2:21 pm

Post by philoum78 » Thu Jan 21, 2010 9:16 pm

TheFug wrote:You can register the account on a ATA or softphone ?
You can also add :5060 behind the sip server uri/address, or when using a sip-uri.
You can't do much more than this, in normal situations you could opt for STUN or proxy servers.
Here is what I can or can't do :
Working :
At the office : Softphone with Direct keyyo account

Not working :
Sipsorcery at the office or home: sipsorcery can't register and put the udp 8060

Softphone at home not working

I guess I'lll have to check the proxy server way, STun does not seem to be recommended.

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TheFug
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Location: The Netherlands, North-Holland

Post by TheFug » Thu Jan 21, 2010 9:36 pm

At work you can't use the 5060 port because they block it ?
by who, or what, are you forced to use port 8060 ? or could this be a config error ?
If you have to use port 8060, you can maybe ask the SipSorcery system operators, to make this port usable at SipSorcery.
Or they can advise you what else to do in this/your case.

Btw. strange, that the softphone option is not working at your home address,
This could be caused by your internet/ADSL/Cable provider,
Do you also make use of phone/voip services by that same provider ?
In most cases the softphone should always work, because transport is via TCP, most of the time and not UDP, SipSorcery Doesn't support TCP btw.
Thanks, The Fug.

gear: my ISP's Zyxel Modem/Router in bridge, Sitecom WL309 Router, Siemens Gigaset 301D

philoum78
Posts: 23
Joined: Mon Jul 06, 2009 2:21 pm

Post by philoum78 » Thu Jan 21, 2010 10:16 pm

TheFug wrote:At work you can't use the 5060 port because they block it ?
by who, or what, are you forced to use port 8060 ? or could this be a config error ?
If you have to use port 8060, you can maybe ask the SipSorcery system operators, to make this port usable at SipSorcery.
Or they can advise you what else to do in this/your case.

Btw. strange, that the softphone option is not working at your home address,
This could be caused by your internet/ADSL/Cable provider,
Do you also make use of phone/voip services by that same provider ?
In most cases the softphone should always work, because transport is via TCP, most of the time and not UDP, SipSorcery Doesn't support TCP btw.
Not easy for me to explain right :oops:

At the office, the softphone x-lite worked after we set the cisco routeur with this:
no ip nat service sip udp port 5060

I try to use sipsorcery as I have other sip account and when I go to the sipsorcery page , Ippi is registar but not keyyo.

it says that :

provider name : IPPI ; owner:XXXXXXXX registar:XXX.XXX.XXX.XXX:5060;last register : today with time
provider name : keyyo ; owner:XXXXXXXX registar:XXX.XXX.XXX.XXX:8060;last register : nothing

So I might have to try to force somewhere in sipsorcery the setting?

At home, I have to change setting of my router and it should work with softphone.

Sorry what I was meaning with softphone is if I try to set the direct link with Keyyo.

My softphone always connect correctly to sipsorcery account :lol:

sorry if I explained wrong

getz.hl@gmail.com
Posts: 5
Joined: Sun Aug 01, 2010 4:37 am

Re: My SIP Switch with your provider

Post by getz.hl@gmail.com » Mon Aug 02, 2010 7:01 am

GBonnet --

I have setup both ATA ports 5060/5061 (two separate landlines) on my SPA2102 with SS and Sipgate. The calls are going through and quality is good. However, both calls are being received as if they originated on port 5060 (appearing to be the same caller, regardless of the origination port).

It looks as though the call is getting reconfigured at Sipgate, where I have the numbers setup as VoIP phones within one account.

[owner]@sipsorcery.com [owner] sip:[owner]@68.xxx.xx.x:5061 01 Aug 2010 23:34:50 3600 udp:68.xxx.xx.x:5061 udp:69.59.142.213:5060 Linksys/SPA2102-3.3.6

How do I enable the call to route back to port 5061? Do I need to configure a second Sipgate one account?

hongkongpom
Posts: 29
Joined: Wed Oct 14, 2009 10:04 am

Re: My SIP Switch with your provider

Post by hongkongpom » Sun Apr 17, 2011 12:03 pm

I can get SS to log onto my Orbtalk and Sipgate accounts with no problem but outgoing calls and using the incoming DID has problems. The phone will ring but we can't hear each other, or I make a call and can't hear the other end ringing.

If I want to make/receive calls from either accounts, I must set my SPA 3102 to bypass SS and connect to the VSP directly which is a pain. This, to me, shows my router isn't the cause as a direct connection by my ATA to the VSP works fine. It's only when I use SS.

I just can't figure this out when SS is only a SIP server and authenticates both ends of the connection. It's almost like if my VSPs don't see an ATA name as registered, they won't allow the calls to go through properly.

eastss
Posts: 6
Joined: Mon Oct 11, 2010 11:19 am

Re: My SIP Switch with your provider

Post by eastss » Sun Apr 24, 2011 10:24 am

I previously saw where sound problems were resolved by adding a port number to
your domain name in your phone setup: You might try adding " :5060 " to the end
of the domain name? Example: sipgate.com:5060

hongkongpom
Posts: 29
Joined: Wed Oct 14, 2009 10:04 am

Re: My SIP Switch with your provider

Post by hongkongpom » Thu Apr 28, 2011 11:20 am

eastss wrote:I previously saw where sound problems were resolved by adding a port number to
your domain name in your phone setup: You might try adding " :5060 " to the end
of the domain name? Example: sipgate.com:5060
Thankyou I tried adding this to the end of the server name but it made no difference.

As soon as I bypass SS the ATA works properly :cry:

hongkongpom
Posts: 29
Joined: Wed Oct 14, 2009 10:04 am

Re: My SIP Switch with your provider

Post by hongkongpom » Wed Nov 02, 2011 11:21 am

I'm wondering if it's because SS to Orbtalk uses a non supported codec? (G711) whereas my SPA3102 is set to G729 and the calls work fine in both directions?

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Aaron
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Re: My SIP Switch with your provider

Post by Aaron » Thu Nov 03, 2011 9:29 am

A codec mismatch is a possibility but that would generally generate a SIP error response before the audio could get established rather than resulting in a call with no audio.

Two more likely scenarios are:

1. Orbtalk doesn't do NAT handling and if your RTP arrives from a socket other than the one specified in your SIP INVITE request then they ignore it. This can happen if your router use an arbitrary port when sending the RTP from your SIP device rather than respecting the sending port number. This scenario would most likely result in no audio at either end of the call.

2. Your router could decide to block the incoming RTP from Orbtalk because it is coming from a socket that it has not previously sent a packet to. It should permit the traffic after it gets an outgoing packet from your SIP device but sometimes routers do weird things. This scenario would most likely result in one-way audio with you not being able to hear anything.

Of course it could be something else completely. Routers on home networks can do weird things when SIP and RTP are involved and are the cause of 99% of the audio issues on SIP calls. If you are able trying a call with a softphone and firing up a packet trace with Wireshark is about the only way to try and dig deeper into what RTP streams are getting through.

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