Greger I am trying to get a cascading hierarchy of geographically separated SIP servers. The reason for this is that I am providing services to a wireless community that connect to the internet via many different gateways to the internet. Most of the users are behind a nat device, and connect to a SER proxy on the wireless net work on the 10.0.0.0/8 network. Each SER server keeps a local location database, but forwards any requests that it cannot complete to a sip proxy on the internet (to which each SER proxy replicates all REGISTER's). It looks a bit like this:
UAC(192.168.0.123)--->WIFI_AP(10.1.2.3)---->WIFI_GATEWAY_1--->INTERNET for browsing / email / ftp etc | | v SER_SERVER on WiFi network (10.0.0.123) also has (public IP Address)--->INTERNET GATEWAY for VOIP 1--->SER_SERVER on internet
^
| UAC(192.168.0.234)--->WIFI_AP(10.3.4.5)---->WIFI_GATEWAY_2--->INTERNET for browsing / email / ftp etc | | | | | v | SER_SERVER on WiFi network (10.3.4.123) | also has (public IP Address)--->INTERNET GATEWAY for VOIP 2-----
Each SER server implements it's own far end nat traversal based on nathelper/rtpproxy, which means that expensive internet links are not used to terminate calls between subscribers on the same wifi network.
The issue I was having with loose routing appeared to be because the SER_SERVER on the WiFi network was multihomed. When a reply cam back from the ser server on the internet (I think it was only ACK's causing the problem), SER tried to loose_route the reply and the next hop was infact the servers own internal ip address.
After loosing much hair, I worked out that if I included the condition I could get the packet to route through to the correct final destination. I am not sure if it's because there is a REAL issue with loose_route not working with mhomed properly, or if it's a peculiarity of my setup.
Noel
Greger V. Teigre wrote:
Noel, It would be interesting to get a description of the setup that caused this problem. Was that Asterisk as well? Same setup as Giovanni? Dependent on auth/non-auth and how Asterisk has been set up with SER, we have seen different problems with signalling and proxying of RTP. I know there was an lr vs. lr=on issue a while back, but somewhere we don't have RFC compliant behavior (all messages with routes should be loose routed). g-)
----- Original Message ----- From: "Noel Sharpe" noels@radnetwork.co.uk To: "Giovanni Balasso" giaso@yahoo.it Cc: serusers@lists.iptel.org Sent: Tuesday, November 15, 2005 10:14 AM Subject: Re: [Serusers] Too Many Hops
sounds to me like you have a looping problem in your script. I had something similar when using the example from OnSIP.org. The loose_route bit needed to be inside a condition: if (uri!=myself){ if (loose_route()) { route(1); }; }; xlog/ngrep is your friend here as you will be able to see which message is being sent between the two servers.
Noel
Giovanni Balasso wrote:
Alle 09:47, mercoledì 02 novembre 2005, Matteo Piazza ha scritto:
I have Ser and asterisk on the same machine. When i try to call with a SIP phone registred on asterisk another sip phone also registred on asterisk through SER I receve this error message: Too many hops
Too many hops is usually reached when there is no rule (or no way) to deliver sip message, adding some log(), or better xlog(), to your routing script could help you (and us) debugging and understanding what's wrong, and which method(s) fail.
if (method == "INVITE") { if (uri =~"^sip:0[0-9]*@*"){ log(1, "Check 1 succed Forwarding to Asterisk\n"); rewritehostport("192.168.9.97:5061"); t_relay(); break; }; };I don't think this will solve your problem but in my experience I had better result with t_relay_to_udp("192.168.9.97","5061") than rewritehostport("192.168.9.97:5061").
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