<html><body><div style="color:#000; background-color:#fff; font-family:HelveticaNeue, Helvetica Neue, Helvetica, Arial, Lucida Grande, sans-serif;font-size:12pt">Yup, that duplicate entry was the problem.<br>Now I have audio but it is always relayed through mediaproxy.<br><br>What I'd like is to have both CSipSimple clients, both using ICE, to try to first establish a direct peer to peer connection to pass the RTP media, and only fallback on mediaproxy if that connection cannot be established. At the moment the CSipSimple logs show that the ICE session never completes. Specifically, the logs show:<br><br>"peer=- (as you can see, no peer IP here at all)<br>ICE role: Unknown state: Candidate Gathering, comp_cnt:2"<br><br>So why isn't ICE completing properly and allowing both peers to establish a direct connection on the LAN (both are on the same LAN behind NAT) instead of relaying automatically through mediaproxy? It seems that automatically realying is an
unnecessary waste of bandwith and only serves to increase latency on the call, specially when the 2 peers can communicate directly which is the case.<br>Is there some configuration in SPCE that I'm overlooking on the ICE side of things to enable what I want?<br><br>Also, can you confirm that indeed there's no support for SIP SIMPLE messaging?<br><br>Thanks<br><div><span><br></span></div><div style="display: block;" class="yahoo_quoted"> <br> <br> <div style="font-family: HelveticaNeue, Helvetica Neue, Helvetica, Arial, Lucida Grande, sans-serif; font-size: 12pt;"> <div style="font-family: HelveticaNeue, Helvetica Neue, Helvetica, Arial, Lucida Grande, sans-serif; font-size: 12pt;"> <div dir="ltr"> <font face="Arial" size="2"> On Thursday, November 21, 2013 12:19 PM, Andrew Pogrebennyk <apogrebennyk@sipwise.com> wrote:<br> </font> </div> <div class="y_msg_container">Hi,<br clear="none">please delete duplicated types from interface lo:<br
clear="none"> - web_ext<br clear="none"> - sip_ext<br clear="none"> - rtp_ext<br clear="none">save, execute ngcpcfg apply and try again.<br clear="none">Not sure how it came to be duplicated, maybe you edited that file manually.<br clear="none"><br clear="none">Usually network.yml is updated automatically on ip address change. This<br clear="none">is done with ip-up.d hook which calls the following command:<br clear="none">ngcp-network --move-from=lo --move-to=eth0 --type=web_ext --type=sip_ext<br clear="none">--type=rtp_ext --type=ssh_ext<br clear="none"><br clear="none">Maybe you had multiple interfaces and the network could not be<br clear="none">auto-configured.<br clear="none"><br clear="none">Andrew<br clear="none"><br clear="none">On 11/21/2013 01:00 PM, Miguel Rios wrote:<br clear="none">> Thanks for your help. Indeed it is listening on eth0 on my public IP,<br
clear="none">> not 0.0.0.0 as I'd stated.<br clear="none">> Here's my network config:<br clear="none">> <br clear="none">> ---<br clear="none">> hosts:<br clear="none">> self:<br clear="none">> interfaces:<br clear="none">> - lo<br clear="none">> - eth0<br clear="none">> <br clear="none">> eth0:<br clear="none">> ip: mypublicIP<br clear="none">> netmask: 255.255.240.0<br clear="none">> type:<br clear="none">> - sip_ext<br clear="none">> - rtp_ext<br clear="none">> - web_ext<br clear="none">> lo:<br clear="none">> ip: 127.0.0.1<br clear="none">> netmask: 255.255.255.0<br clear="none">> shared_ip:
[]<br clear="none">> shared_v6ip: []<br clear="none">> type:<br clear="none">> - sip_int<br clear="none">> - ha_int<br clear="none">> - web_ext<br clear="none">> - sip_ext<br clear="none">> - rtp_ext<br clear="none">> - web_int<br clear="none">> v6ip: '::1'<br clear="none">> role:<br clear="none">> - proxy<br clear="none">> - lb<br clear="none">> - mgmt<br clear="none">> <br clear="none">> <br clear="none">> Here's my netstat output. I added port 5071 manually as per the<br clear="none">> instructions, and indeed I can register TLS port 5071 from my CSipSimple<br clear="none">> client:<br
clear="none">> <br clear="none">> 12:listen=udp:127.0.0.1:5060<br clear="none">> 13:listen=tcp:127.0.0.1:5060<br clear="none">> 15:listen=tls:127.0.0.1:5071<br clear="none">> 17:listen=udp:mypublicip:5060<br clear="none">> 18:listen=tcp:mypublicip:5060<br clear="none">> 20:listen=tls:mypublicip:5071<br clear="none">> 23:listen=udp:[::1]:5060<br clear="none">> 24:listen=tcp:[::1]:5060<br clear="none">> 26:listen=tls:[::1]:5071<br clear="none">> 29:# additional sockets to listen on, if any<br clear="none">> 33:listen=udp:127.0.0.1:5060<br clear="none">> 35:listen=tcp:127.0.0.1:5060<br clear="none">> <br clear="none">> <br clear="none">> So it seems that it is properly configured, no? Perhaps the problem lies<br clear="none">> with ICE negotiation in CSipSimple...<br clear="none">> How does spce deal with ICE on the server side of things?<br clear="none">> <br clear="none">> Thanks<br
clear="none">> <br clear="none">> <br clear="none">> On Thursday, November 21, 2013 11:43 AM, Daniel Grotti<br clear="none">> <<a shape="rect" ymailto="mailto:dgrotti@sipwise.com" href="mailto:dgrotti@sipwise.com">dgrotti@sipwise.com</a>> wrote:<br clear="none">> Hi,<br clear="none">> kamailio must run on eth0 (usually), not on 0.0.0.0.<br clear="none">> You should have something like:<br clear="none">> <br clear="none">> hosts:<br clear="none">> self:<br clear="none">> eth0:<br clear="none">> ip: 10.15.20.110<br clear="none">> netmask: 255.255.255.0<br clear="none">> type:<br clear="none">> - web_ext<br clear="none">> - sip_ext<br clear="none">> - rtp_ext<br clear="none">> - ssh_ext<br
clear="none">> interfaces:<br clear="none">> - lo<br clear="none">> - eth0<br clear="none">> lo:<br clear="none">> ip: 127.0.0.1<br clear="none">> netmask: 255.255.255.0<br clear="none">> shared_ip: []<br clear="none">> shared_v6ip: []<br clear="none">> type:<br clear="none">> - sip_int<br clear="none">> - ha_int<br clear="none">> - web_int<br clear="none">> v6ip: '::1'<br clear="none">> role:<br clear="none">> - proxy<br clear="none">> - lb<br clear="none">> - mgmt<br clear="none">> <br clear="none">> <br clear="none">> <br clear="none">>
That ip will be use also from mediaproxy to performa mediarelay.<br clear="none">> So LB will listen on:<br clear="none">> <br clear="none">> <a shape="rect" ymailto="mailto:root@spce" href="mailto:root@spce">root@spce</a> <mailto:<a shape="rect" ymailto="mailto:root@spce" href="mailto:root@spce">root@spce</a>>:~# grep -nr "listen"<br clear="none">> /etc/kamailio/lb/kamailio.cfg<br clear="none">> 12:listen=udp:10.15.20.110:5060<br clear="none">> 13:listen=tcp:10.15.20.110:5060<br clear="none">> 15:listen=tls:10.15.20.110:5061<br clear="none">> 19:# additional sockets to listen on, if any<br clear="none">> 23:listen=udp:127.0.0.1:5060<br clear="none">> 25:listen=tcp:127.0.0.1:5060<br clear="none">> <br clear="none">> <br clear="none">> <br clear="none">> Daniel<br clear="none">> <br clear="none">> <br clear="none">> <br clear="none">> On 11/21/2013 12:31 PM, Miguel Rios wrote:<br
clear="none">>> Hi,<br clear="none">>><br clear="none">>> I've set up a simple spce v 3.0 install on debian 7 but I'm experiencing<br clear="none">>> no audio.<br clear="none">>><br clear="none">>> When I ran the setup script I noticed that my kamailio instance was only<br clear="none">>> listening on localhost.<br clear="none">>> I changed that in network.yml and now kamailio listens on 0.0.0.0 and my<br clear="none">>> CSipSimple clients can register OK.<br clear="none">>><br clear="none">>> However, when I have 2 registered CSipSimple clients try to call each<br clear="none">>> other, I can answer but there's no audio.<br clear="none">>><br clear="none">>> I see that CSipSimple log shows ICE negotiation in progress but it never<br clear="none">>> concludes. Now given that the 2 clients were on the same LAN behind NAT,<br clear="none">>> shouldn't they have
negotiated a media path directly between them with<br clear="none">>> their private IPs?<br clear="none">>><br clear="none">>> Since the spce setup script ended up putting kamailio listening only on<br clear="none">>> localhost, could the same have happened to media proxy? Where do I need<br clear="none">>> to check to make sure the install correctly configured the interfaces?<br clear="none">>><br clear="none">>> Also, what are the recommended settings to use with CSipSimple and spce<br clear="none">>> (ICE/STUN/TURN wise)? I have ICE enabled on CSipSimple and I thought<br clear="none">>> that would be enough, no?<br clear="none">>><br clear="none">>> Lastly, I tried sending an IM through CSipSimple, which uses SIP SIMPLE,<br clear="none">>> but I got an error. Does spce not support SIP SIMPLE messaging?<br clear="none">>><br clear="none">>> Thanks<br
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