Hello! Yesterday I've successfully placed a call between two different domains: originate sofia/default/1003@10.99.8.221 &bridge(sofia/gateway/212.235.180.41/1001) I didn't hear any audio, but it was kinda working. Today I investigated this more deep and found some issues. FS with 212.235.180.41 is a public computer with firewall, but open TCP and UDP 5060, 5080 ports. Freeswitch on this machine uses default configuration. FS with 10.99.8.221 is a lan computer in a different place, this is where I would like to start a call, the other way would be probably too much difficult for now. I've added a gateway entry to this one: http://pastebin.com/m2174ead Calling from 10.99.8.221 (for e.g. using softphone at ext. 1003) to 212.235.180.41 (ext. 1001 for e.g.) works. Both end answers, however I cannot hear audio coming trough. When testing I'm at the computer which is behind a lan, so I'm capturing music as audio source on the other side. Are there any other ports I should open on public computer? With wireshark on the computer behind a lan, I can see RTP going away to 212.235.180.41, but not the other way. There are also issues when e.g. terminating a call on public computer, fs on the other end will never terminate the call since SIP messages cannot reach the computer behind lan I guess, but this is second problem. -- kriko
kriko <kristjan.ugrin@gmail.com> wrote:
Are there any other ports I should open on public computer? With wireshark on the computer behind a lan, I can see RTP going away to 212.235.180.41, but not the other way.
Maybe the NAT device between the two machines is blocking the rtp traffic. Can you configure the NAT device to forward incoming rtp to the correct destination on the LAN? If you capture packets on the machine with the public IP address and it shows the RTP traffic being sent, this is evidence that the NAT device in between is causing your problems Have a look also at the wiki pages related to NAT.
There are also issues when e.g. terminating a call on public computer, fs on the other end will never terminate the call since SIP messages cannot reach the computer behind lan I guess, but this is second problem.
This is fixed by having SIP packets (port 6080 in the default external profile) forwarded properly by the NAT device to the machine on the LAN. In my router's configuration: ip nat source static udp 192.168.0.2 5080 interface Dialer1 5080 I don't need to worry about the RTP ports because IP inspection is enabled, and it seems to handle everything.
Generally speaking you will need to open an UPD port range for the RTP stream. This can be configured on FS. Eg. we use 12000-13000 on our system. Then If you do not hear any sound you may put <param name="ext-rtp-ip" value="stun:stun.freeswitch.org"/> in your external and internal profile, if FS is natted. Best regards Peter kriko schrieb:
Hello!
Yesterday I've successfully placed a call between two different domains: originate sofia/default/1003@10.99.8.221 &bridge(sofia/gateway/212.235.180.41/1001)
I didn't hear any audio, but it was kinda working. Today I investigated this more deep and found some issues. FS with 212.235.180.41 is a public computer with firewall, but open TCP and UDP 5060, 5080 ports. Freeswitch on this machine uses default configuration.
FS with 10.99.8.221 is a lan computer in a different place, this is where I would like to start a call, the other way would be probably too much difficult for now. I've added a gateway entry to this one: http://pastebin.com/m2174ead
Calling from 10.99.8.221 (for e.g. using softphone at ext. 1003) to 212.235.180.41 (ext. 1001 for e.g.) works. Both end answers, however I cannot hear audio coming trough. When testing I'm at the computer which is behind a lan, so I'm capturing music as audio source on the other side.
Are there any other ports I should open on public computer? With wireshark on the computer behind a lan, I can see RTP going away to 212.235.180.41, but not the other way.
There are also issues when e.g. terminating a call on public computer, fs on the other end will never terminate the call since SIP messages cannot reach the computer behind lan I guess, but this is second problem.
Thanks for all suggestions. Ufortunately I cannot get it working. Seems like packets are not coming to phone behind nat (freeswitch is on public ip). When registering I can see multiple notify retries like this: send 802 bytes to udp/[10.99.10.6]:5060 at 10:49:31.762605: ------------------------------------------------------------------------ NOTIFY sip:1003@10.99.10.6;transport=UDP SIP/2.0 Via: SIP/2.0/UDP 212.235.180.41:5080;rport;branch=z9hG4bKtNStS2gtr8DNr Max-Forwards: 70 From: <sip:1003@212.235.180.41>;tag=veSr4DmgmFHjr To: <sip:1003@212.235.180.41> Call-ID: cec2b00b-5814-122c-f981-000fea488302 CSeq: 109587536 NOTIFY Contact: <sip:mod_sofia@212.235.180.41:5080> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-10924M Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO Supported: timer, precondition, path, replaces Event: message-summary Allow-Events: talk, refer Subscription-State: terminated;timeout Content-Type: application/simple-message-summary Content-Length: 93 Messages-Waiting: yes Message-Account: sip:1003@212.235.180.41 Voice-Message: 3/0 (0/0) I've opened necessary ports and I've defined custom rtp port range (which goes trough). Does nat should really just work if you register on external profile via port 5080? This is what I'm doing now. The phone on lan is a nokia N95 configured like described here (using port 5080): http://wiki.freeswitch.org/wiki/Nokia_N95 Phone shows registered message, but it takes like a half minute to register, when I'm on home network this happens in a second. On Wed, 07 Jan 2009 12:39:45 +0100, Peter P GMX <Prometheus001@gmx.net> wrote:
Generally speaking you will need to open an UPD port range for the RTP stream. This can be configured on FS. Eg. we use 12000-13000 on our system. Then If you do not hear any sound you may put
<param name="ext-rtp-ip" value="stun:stun.freeswitch.org"/>
in your external and internal profile, if FS is natted.
Best regards Peter
kriko schrieb:
Hello!
Yesterday I've successfully placed a call between two different domains: originate sofia/default/1003@10.99.8.221 &bridge(sofia/gateway/212.235.180.41/1001)
I didn't hear any audio, but it was kinda working. Today I investigated this more deep and found some issues. FS with 212.235.180.41 is a public computer with firewall, but open TCP and UDP 5060, 5080 ports. Freeswitch on this machine uses default configuration.
FS with 10.99.8.221 is a lan computer in a different place, this is where I would like to start a call, the other way would be probably too much difficult for now. I've added a gateway entry to this one: http://pastebin.com/m2174ead
Calling from 10.99.8.221 (for e.g. using softphone at ext. 1003) to 212.235.180.41 (ext. 1001 for e.g.) works. Both end answers, however I cannot hear audio coming trough. When testing I'm at the computer which is behind a lan, so I'm capturing music as audio source on the other side.
Are there any other ports I should open on public computer? With wireshark on the computer behind a lan, I can see RTP going away to 212.235.180.41, but not the other way.
There are also issues when e.g. terminating a call on public computer, fs on the other end will never terminate the call since SIP messages cannot reach the computer behind lan I guess, but this is second problem.
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-- kriko
participants (3)
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Jason White
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kriko
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Peter P GMX