Re: [asterisk-users] Differences between Chan_SIP and PJSIP with NAT and STUN
>> check the system and make sure there really is no firewall like I said > You were right. Stick around on the list long enough and you'll realise the truth... he always is ;-) Pete -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Differences between Chan_SIP and PJSIP with NAT and STUN
You were right. I had non-default rtp ports open in iptables. Edited rtp.conf et voila. Everything seems to be working. Thanks so much for your patience and guidance! Have a lovely eening. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Differences between Chan_SIP and PJSIP with NAT and STUN
Chirag Desai wrote: So I see: EXTERNAL_SNOM_IP -> EXTERNAL_ASTERISK_IP (UDP, length 218, src: 60798, dst 11128) EXTERNAL_ASTERISK_IP -> INTERNAL_SNOM_IP (UDP, length 218, src: 11128 dst 60478 So i see udp from the phone, but there's no audio. If "rtp set debug on" shows no packets being received then they are not being read off the socket, so I'd check the system and make sure there really is no firewall like I said. Once packets start getting received then we'll change the target address and audio will flow. It may even be that when using TCP or UDP you have some sort of helper which is opening up the right firewall ports and when TLS is in use it can't see the traffic and thus doesn't. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Differences between Chan_SIP and PJSIP with NAT and STUN
So I see: EXTERNAL_SNOM_IP -> EXTERNAL_ASTERISK_IP (UDP, length 218, src: 60798, dst 11128) EXTERNAL_ASTERISK_IP -> INTERNAL_SNOM_IP (UDP, length 218, src: 11128 dst 60478 So i see udp from the phone, but there's no audio. I do also see some packets :: EXTERNAL_ASTERISK_IP -> EXTERNAL_SNOM_IP (ICMP, length 246, Destination unreachable (Host administratively prohibited) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Differences between Chan_SIP and PJSIP with NAT and STUN
Chirag Desai wrote: In the PCAP I can see asterisk sending UDP packets to my local IP 192.168.0.5 If you don't see anything arriving from the remote side and we've told them the right IP address and ICE is not actually negotiated... then that leans more towards something remote unless there actually is a firewall. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Differences between Chan_SIP and PJSIP with NAT and STUN
In the PCAP I can see asterisk sending UDP packets to my local IP 192.168.0.5 It's funny, when I switch to TCP on 5060 audio seems to work fine. The moment I go to 5063 on TLS everything goes a bit awry. Any further input is greatly appreciated. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Differences between Chan_SIP and PJSIP with NAT and STUN
Chirag Desai wrote: I'm dialling from the snom and every few calls asterisk sends media to the phones external IP and it works! And then now and again it sends the media to the phones internal IP and I hear nothing. I'm really at a loss. In the non-working case check the IP address in the SDP, if it's the external then we've told the phone to send it to the right place. After that do a packet capture and see if the packets are arriving on the machine. If not then look outside the machine at things. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Differences between Chan_SIP and PJSIP with NAT and STUN
I'm dialling from the snom and every few calls asterisk sends media to the phones external IP and it works! And then now and again it sends the media to the phones internal IP and I hear nothing. I'm really at a loss. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Differences between Chan_SIP and PJSIP with NAT and STUN
Chirag Desai wrote: Joshua Colp wrote: Have you done a packet capture to see if the RTP from the remote device is hitting the machine to narrow things down? Nope. When I run with RTP encryption on it seems that rewrite_contact does not work in PJSIP. When I turn off RTP some calls get media, some don't. If you look at the SIP trace it seems like the rewrite_contact doesn't always take affect. The rewrite_contact shows as working fine in the SIP trace. The log shows the message as received over the socket, before modification. If it wasn't working then the BYE would be going to the internal IP address+port. Nothing stands out in the signaling. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Differences between Chan_SIP and PJSIP with NAT and STUN
> Joshua Colp wrote: >> >> Have you done a packet capture to see if the RTP from the remote device >> is hitting the machine to narrow things down? >> >> >> Nope. When I run with RTP encryption on it seems that rewrite_contact does not work in PJSIP. When I turn off RTP some calls get media, some don't. If you look at the SIP trace it seems like the rewrite_contact doesn't always take affect. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Differences between Chan_SIP and PJSIP with NAT and STUN
> Joshua Colp wrote: > > There should be nothing different, except for how you configure things. > What is the full PJSIP configuration? What is the environment where > Asterisk is running? Is ICE actually in use on the other side? What is > the full SIP trace? > The full configuration is here: http://pastebin.com/XqZG1m5X I am connection over TLS / SRTP on port 5063. When I put in a stun server asterisk sends media to the phone's external IP. The asterisk is has a public IP and internal IP. It is internet facing, and is not behind NAT. When I had ICE enabled on the snom, it didnt seem to make any difference. PJ showed an ICE error. The sip trace is here: http://pastebin.com/fDxbk289 Thanks for your help. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Differences between Chan_SIP and PJSIP with NAT and STUN
Chirag Desai wrote: I have two accounts on Asterisk 13. One uses chan_sip and the other pjsip. In my snom 760 the setup for these two accounts is identical. When I call echo test from the account using chan_sip audio comes through fine. When I call echo test from the account using pjsip there is no audio. With rtp set debug on, I can see that audio is being sent to the snom's internal IP 192.168.0.x I can add a stun server in the config for this account and RTP flows to the Public IP and I get audio. I was wondering why there is a difference between pjsip and chan_sip so that one works without stun and the other requires it. Does anybody know why? Maybe my settings are off in pjsip. There should be nothing different, except for how you configure things. What is the full PJSIP configuration? What is the environment where Asterisk is running? Is ICE actually in use on the other side? What is the full SIP trace? -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
