esojmc
join:2002-07-08
Naperville, IL
| TrixBox: Question on multiple SIP trunks
I have TB setup at home and I am connecting to two SIP providers, ViaTalk and SipPhone.com. I was looking through my new Tomato enable router and noticed that my TB was connecting to both SIP providers with source port 5060. Is that OK? Reason I ask is that there is always a 3-4 second delay on incoming calls via ViaTalk. For example, when a call comes in and I pick up the phone and say hello, the calling party can't hear me say hello. I have to repeat the hello before they can hear me. I was hoping this would explain the delay.
I have 5060-5080 and 10000-20000 forwarded to my TB.
Thanks |
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cbrain
join:2000-05-21
Silver Spring, MD
 ·VoicePulse for Bus..
 ·Verizon FIOS
 ·Comcast
·Future Nine Corpor..
·VoiceStick
| said by esojmc  :... I am connecting to two SIP providers, ViaTalk and SipPhone.com. ... Is the delay always ViaTalk and never SipPhone? If you have doubt, disable SipPhone while you test ViaTalk.
It shouldn't cause the problem. |
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tommy13v
Premium
join:2002-02-15
Glenville NY | reply to esojmc
It's not a problem. |
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esojmc
join:2002-07-08
Naperville, IL
| I am just trying to understand how both services can use 5060 as the source port. I haven't tried this but what happens when I have a call in progress on both trunks? Wouldn't all the data go into my TB's 5060 port. How would TB differentiate the two separate calls?
@cbrain
Good suggestion - I will try removing the sipphone.com trunk later today and see what happens. I should have thought about doing that.
Thanks for the replies. |
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tommy13v
Premium
join:2002-02-15
Glenville NY
·ViaTalk
| UDP Port 5060 is for SIP communication. This is only used for setting up calls, taking down calls and so on. Once the call is setup it negotiates the RTP ports to use in the range of 10000-20000 for voice communication. Once the call ends it uses 5060 from your server to send the signal to your provider to hangup the call.
Take a look over at »www.voip-info.org for more explanations. |
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cbrain
join:2000-05-21
Silver Spring, MD | reply to esojmc
Magic computers and routers. Haven't you seen a large office with 100's surfing the web on port 80?  |
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esojmc
join:2002-07-08
Naperville, IL
| reply to cbrain
I did some experimenting and it looks like a problem with TrixBox and ViaTalk. Here is what I did:
- Removed the SipPhone.com trunk but there was still a 3-4 second delay on incoming calls to ViaTalk. There is no delay on my cell phone when I use simultaneous ring.
- Disconnected TB and connected to ViaTalk using my PAP2. No delay on incoming calls.
This leads me to believe that some interaction between VT and TB is causing the delay on incoming calls. Outgoing calls are fine.
Is anyone using TrixBox and ViaTalk without this problem? I sure would like to know how you have it configured.
Thanks |
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username1961
Premium
join:2007-08-01
North Royalton, OH
1 edit | I use callweaver (asterisk) with viatalk and noticed the same problem. I think it is viatalk. I don't see the problem with broadvice, sunrocket (before they closed), voipyourlife or siphone.
Trixbox has to use the same port 5060. This is the SIP port. The call get routed based on the "To:" field in the SIP packet not the port. |
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cbrain
join:2000-05-21
Silver Spring, MD
·VoicePulse for Bus..
·Verizon FIOS
·Comcast
·Future Nine Corpor..
·VoiceStick
| said by username1961 :I use callweaver (asterisk) with viatalk and noticed the same problem. I think it is viatalk. I don't see the problem with broadvice, sunrocket (before they closed), voipyourlife or siphone. Trixbox has to use the same port 5060. This is the SIP port. The call get routed based on the "To:" field in the SIP packet not the port. Tell us about Callweaver please. |
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tommy13v
Premium
join:2002-02-15
Glenville NY
1 edit | reply to esojmc
I am not having this issue with plain Asterisk. I am connecting to the chicago servers.
Are you recording the calls?
Have you put in your externip in sip.conf?
Maybe do a sip debug on the incoming call. |
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esojmc
join:2002-07-08
Naperville, IL | reply to esojmc
Found a solution for the 2-3 second delay on incoming call audio.
»Viatalk & Asterisk - fill me in |
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priller
join:2000-10-20
Gainesville, VA
·voip.ms
·Callcentric
·Vonage
·callwithus
| reply to esojmc
said by esojmc :I am just trying to understand how both services can use 5060 as the source port. I haven't tried this but what happens when I have a call in progress on both trunks? Wouldn't all the data go into my TB's 5060 port. How would TB differentiate the two separate calls? The piece you're missing is that it's the port plus the source and destination IP address ... they're going to be different and that's what makes up the router's translation table.
On another note, if you have a router that requires you to forward ports to make VoIP work, then it's time to get a new router ... honestly. |
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