Call between gtalk and sip - no audio
I modified mod_dingaling.c so I can intercept google talk chat messages via socket - nothing fancy. Then I wrote a java app that connects to freeswitch socket and in case of a proper message (trigger) it sends a command to freeswitch, in my case: api originate sofia/default/1001@10.99.8.221 &bridge(dingaling/gmail.com/my_mail@gmail.com) Dingaling is logged in as another user which I have added as buddy, chat messages go trough, however when a call is started between SIP and Gtalk client, we cannot hear each other at all. Using freeswitch revision: 10866 Here is the log: http://pastebin.com/m1eba2cb8 What can be the problem? First I thought it is because running sip client + gtalk and freeswitch on one host, but then I moved SIP phone and Gtalk to 2 different workstations, using the third only for freeswitch. Also calls from "call" example program from google lib works fine with same setup - something must be problematic with freeswitch, however cannot see what. Thank you! -- kriko
Your log shows rtp streams being allocated. did you look at at the packets on the wire with a packet capture program? You are better off using proper jingle and component mode. What you are describing sounds like a workaround to avoid doing it right. On Mon, Dec 22, 2008 at 8:42 AM, kriko <kristjan.ugrin@gmail.com> wrote:
I modified mod_dingaling.c so I can intercept google talk chat messages via socket - nothing fancy. Then I wrote a java app that connects to freeswitch socket and in case of a proper message (trigger) it sends a command to freeswitch, in my case: api originate sofia/default/1001@10.99.8.221 &bridge(dingaling/gmail.com/my_mail@gmail.com)
Dingaling is logged in as another user which I have added as buddy, chat messages go trough, however when a call is started between SIP and Gtalk client, we cannot hear each other at all. Using freeswitch revision: 10866
Here is the log: http://pastebin.com/m1eba2cb8
What can be the problem? First I thought it is because running sip client + gtalk and freeswitch on one host, but then I moved SIP phone and Gtalk to 2 different workstations, using the third only for freeswitch. Also calls from "call" example program from google lib works fine with same setup - something must be problematic with freeswitch, however cannot see what.
Thank you!
-- kriko
_______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
There are absolutely no UDP packets going trough like when doing a call from gtalk to gtalk. You mean this (component mode): http://wiki.freeswitch.org/wiki/Dingaling#What_is_Component_.28server_to_ser... Is there more documentation that this? All I would like to do is to initiate a call between SIP telephone and gtalk user who typed in the message. Thank you! On Mon, 22 Dec 2008 16:19:08 +0100, Anthony Minessale <anthony.minessale@gmail.com> wrote:
Your log shows rtp streams being allocated. did you look at at the packets on the wire with a packet capture program?
You are better off using proper jingle and component mode. What you are describing sounds like a workaround to avoid doing it right.
On Mon, Dec 22, 2008 at 8:42 AM, kriko <kristjan.ugrin@gmail.com> wrote:
I modified mod_dingaling.c so I can intercept google talk chat messages via socket - nothing fancy. Then I wrote a java app that connects to freeswitch socket and in case of a proper message (trigger) it sends a command to freeswitch, in my case: api originate sofia/default/1001@10.99.8.221 &bridge(dingaling/gmail.com/my_mail@gmail.com)
Dingaling is logged in as another user which I have added as buddy, chat messages go trough, however when a call is started between SIP and Gtalk client, we cannot hear each other at all. Using freeswitch revision: 10866
Here is the log: http://pastebin.com/m1eba2cb8
What can be the problem? First I thought it is because running sip client + gtalk and freeswitch on one host, but then I moved SIP phone and Gtalk to 2 different workstations, using the third only for freeswitch. Also calls from "call" example program from google lib works fine with same setup - something must be problematic with freeswitch, however cannot see what.
Thank you!
-- kriko
_______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Porn - the reason you need a new hard drive.
are you doing the trace on the FS box too? it says it's established RTP and bridging. NO audio is 9.8/10 times a firewall issue. typing in a message is not the right way to call someone on jingle. That's the point. In component mode you add the sip ext to your buddy list and call them the normal way. This has nothing to do with your audio issue though so it's not a big deal. On Mon, Dec 22, 2008 at 9:42 AM, kriko <kristjan.ugrin@gmail.com> wrote:
There are absolutely no UDP packets going trough like when doing a call from gtalk to gtalk.
You mean this (component mode):
http://wiki.freeswitch.org/wiki/Dingaling#What_is_Component_.28server_to_ser... Is there more documentation that this?
All I would like to do is to initiate a call between SIP telephone and gtalk user who typed in the message.
Thank you!
On Mon, 22 Dec 2008 16:19:08 +0100, Anthony Minessale <anthony.minessale@gmail.com> wrote:
Your log shows rtp streams being allocated. did you look at at the packets on the wire with a packet capture program?
You are better off using proper jingle and component mode. What you are describing sounds like a workaround to avoid doing it right.
On Mon, Dec 22, 2008 at 8:42 AM, kriko <kristjan.ugrin@gmail.com> wrote:
I modified mod_dingaling.c so I can intercept google talk chat messages via socket - nothing fancy. Then I wrote a java app that connects to freeswitch socket and in case of a proper message (trigger) it sends a command to freeswitch, in my case: api originate sofia/default/1001@10.99.8.221 &bridge(dingaling/gmail.com/my_mail@gmail.com)
Dingaling is logged in as another user which I have added as buddy, chat messages go trough, however when a call is started between SIP and Gtalk client, we cannot hear each other at all. Using freeswitch revision: 10866
Here is the log: http://pastebin.com/m1eba2cb8
What can be the problem? First I thought it is because running sip client + gtalk and freeswitch on one host, but then I moved SIP phone and Gtalk to 2 different workstations, using the third only for freeswitch. Also calls from "call" example program from google lib works fine with same setup - something must be problematic with freeswitch, however cannot see what.
Thank you!
-- kriko
_______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Porn - the reason you need a new hard drive.
_______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
But what I would like to achieve is something different (quite similar). You type in a message like "call 1001@10.99.8.20" and you it would call a SIP buddy with any local number. In component mode you need to add a buddy everytime for a different sip nr.? Which would mean a lot of numbers if you would like to call more than one sip nr. in a lan. As for the first issue, there are RTP packets traveling on FS, but never reach destination after they leave our internal network. Do they have to go outside lan even when the call is placed in a lan between gtalk and SIP? Gtalk to gtalk is no problem on same machines... On Mon, 22 Dec 2008 17:02:02 +0100, Anthony Minessale <anthony.minessale@gmail.com> wrote:
are you doing the trace on the FS box too? it says it's established RTP and bridging.
NO audio is 9.8/10 times a firewall issue.
typing in a message is not the right way to call someone on jingle. That's the point. In component mode you add the sip ext to your buddy list and call them the normal way. This has nothing to do with your audio issue though so it's not a big deal.
On Mon, Dec 22, 2008 at 9:42 AM, kriko <kristjan.ugrin@gmail.com> wrote:
There are absolutely no UDP packets going trough like when doing a call from gtalk to gtalk.
You mean this (component mode):
http://wiki.freeswitch.org/wiki/Dingaling#What_is_Component_.28server_to_ser... Is there more documentation that this?
All I would like to do is to initiate a call between SIP telephone and gtalk user who typed in the message.
Thank you!
On Mon, 22 Dec 2008 16:19:08 +0100, Anthony Minessale <anthony.minessale@gmail.com> wrote:
Your log shows rtp streams being allocated. did you look at at the packets on the wire with a packet capture program?
You are better off using proper jingle and component mode. What you are describing sounds like a workaround to avoid doing it right.
On Mon, Dec 22, 2008 at 8:42 AM, kriko <kristjan.ugrin@gmail.com> wrote:
I modified mod_dingaling.c so I can intercept google talk chat messages via socket - nothing fancy. Then I wrote a java app that connects to freeswitch socket and in case of a proper message (trigger) it sends a command to freeswitch, in my case: api originate sofia/default/1001@10.99.8.221 &bridge(dingaling/gmail.com/my_mail@gmail.com)
Dingaling is logged in as another user which I have added as buddy, chat messages go trough, however when a call is started between SIP and Gtalk client, we cannot hear each other at all. Using freeswitch revision: 10866
Here is the log: http://pastebin.com/m1eba2cb8
What can be the problem? First I thought it is because running sip client + gtalk and freeswitch on one host, but then I moved SIP phone and Gtalk to 2 different workstations, using the third only for freeswitch. Also calls from "call" example program from google lib works fine with same setup - something must be problematic with freeswitch, however cannot see what.
Thank you!
-- kriko
_______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Porn - the reason you need a new hard drive.
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--
if you see them leave FS and never reach dest. It implies a firewall somewhere in between is blocking them. On Mon, Dec 22, 2008 at 10:19 AM, kriko <kristjan.ugrin@gmail.com> wrote:
But what I would like to achieve is something different (quite similar). You type in a message like "call 1001@10.99.8.20" and you it would call a SIP buddy with any local number.
In component mode you need to add a buddy everytime for a different sip nr.? Which would mean a lot of numbers if you would like to call more than one sip nr. in a lan.
As for the first issue, there are RTP packets traveling on FS, but never reach destination after they leave our internal network. Do they have to go outside lan even when the call is placed in a lan between gtalk and SIP? Gtalk to gtalk is no problem on same machines...
On Mon, 22 Dec 2008 17:02:02 +0100, Anthony Minessale <anthony.minessale@gmail.com> wrote:
are you doing the trace on the FS box too? it says it's established RTP and bridging.
NO audio is 9.8/10 times a firewall issue.
typing in a message is not the right way to call someone on jingle. That's the point. In component mode you add the sip ext to your buddy list and call them the normal way. This has nothing to do with your audio issue though so it's not a big deal.
On Mon, Dec 22, 2008 at 9:42 AM, kriko <kristjan.ugrin@gmail.com> wrote:
There are absolutely no UDP packets going trough like when doing a call from gtalk to gtalk.
You mean this (component mode):
http://wiki.freeswitch.org/wiki/Dingaling#What_is_Component_.28server_to_ser...
Is there more documentation that this?
All I would like to do is to initiate a call between SIP telephone and gtalk user who typed in the message.
Thank you!
On Mon, 22 Dec 2008 16:19:08 +0100, Anthony Minessale <anthony.minessale@gmail.com> wrote:
Your log shows rtp streams being allocated. did you look at at the packets on the wire with a packet capture program?
You are better off using proper jingle and component mode. What you are describing sounds like a workaround to avoid doing it right.
On Mon, Dec 22, 2008 at 8:42 AM, kriko <kristjan.ugrin@gmail.com> wrote:
I modified mod_dingaling.c so I can intercept google talk chat messages via socket - nothing fancy. Then I wrote a java app that connects to freeswitch socket and in case of a proper message (trigger) it sends a command to freeswitch, in my case: api originate sofia/default/1001@10.99.8.221 &bridge(dingaling/gmail.com/my_mail@gmail.com)
Dingaling is logged in as another user which I have added as buddy, chat messages go trough, however when a call is started between SIP and Gtalk client, we cannot hear each other at all. Using freeswitch revision: 10866
Here is the log: http://pastebin.com/m1eba2cb8
What can be the problem? First I thought it is because running sip client + gtalk and freeswitch on one host, but then I moved SIP phone and Gtalk to 2 different workstations, using the third only for freeswitch. Also calls from "call" example program from google lib works fine with same setup - something must be problematic with freeswitch, however cannot see what.
Thank you!
-- kriko
_______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Porn - the reason you need a new hard drive.
_______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
--
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-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
Today I did some more testing and packet sniffing. When calling from google talk to google talk, packets are traveling only inside lan, there are some queries which goes outside, but nothing more. When using Gtalk to call someone on sip, then those packets are sent outside and I never see them again. I think this is freeswitch configuration problem (routing?). Where can I look further to investigate why this happens? Thanks. On Mon, 22 Dec 2008 18:30:46 +0100, Anthony Minessale <anthony.minessale@gmail.com> wrote:
if you see them leave FS and never reach dest. It implies a firewall somewhere in between is blocking them.
On Mon, Dec 22, 2008 at 10:19 AM, kriko <kristjan.ugrin@gmail.com> wrote:
But what I would like to achieve is something different (quite similar). You type in a message like "call 1001@10.99.8.20" and you it would call a SIP buddy with any local number.
In component mode you need to add a buddy everytime for a different sip nr.? Which would mean a lot of numbers if you would like to call more than one sip nr. in a lan.
As for the first issue, there are RTP packets traveling on FS, but never reach destination after they leave our internal network. Do they have to go outside lan even when the call is placed in a lan between gtalk and SIP? Gtalk to gtalk is no problem on same machines...
_______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
I've decided to do this properly: clean fresweetch reinstall. My worsktation hosts freeswitch + 1 sip phone also running as 1000 (linux - IP 10.99.8.221) Other windows machine has gtalk with and also a sip phone registered as 1001 (IP 10.99.8.111). First case - SIP to SIP. Calling from 1000 to 1001 and vice versa works, audio is perfect. Packets are propery travelling between 10.99.8.221 and 10.99.8.111 Second case : On windows machine I open gtalk and I open a chat to buddy which is actually a bot logged in on freeswitch (dingaling client mode). The I started java socket program which listens to icoming messages, after typing into client "call 1000@10.99.8.221" an api command is executed: "api originate sofia/default/1000@10.99.8.221 &bridge(dingaling/gmail.com/gtalk_mail(at)gmail.com)" A call is placed between gtalk and sip phone 1000, it rings, but when both end answers there is no audio. After a minute, the call ends itself. I've attached wireshark dumps from both ends - what is strange is that packets are not trying to got at right IP, instead they hit some other machine (213.x.x.x), which doesn't make sense. Fresh log from freeswitch (I don't know why 213.x.x.x gets mixed in this story): http://pastebin.com/m75b10388 // I hope the attachments go trough - 17 KB. test_gtalk_client_side - dump from win machine (gtalk client) test_sip_client - dump from linux machine (freeswitch and sip phone client) I hope to get resolved this mistery somehow. Thank you for all kind answers.
when 2 devices talk via googles gtalk when they are both behind the same lan you are going to have problems. on thing you can do is make an acl to ignore any candidates that are not local add this to your dingaling profile <param name="candidate-acl" value="myacl"/> then add myacl to acl.conf.xml that only allows your lan ip. Turn off all the stun and ext-rtp-ip setting. OR use the windows machine from a box that is not on the sam lan behind the same nat. On Tue, Dec 23, 2008 at 8:09 AM, kriko <kristjan.ugrin@gmail.com> wrote:
I've decided to do this properly: clean fresweetch reinstall.
My worsktation hosts freeswitch + 1 sip phone also running as 1000 (linux - IP 10.99.8.221) Other windows machine has gtalk with and also a sip phone registered as 1001 (IP 10.99.8.111).
First case - SIP to SIP. Calling from 1000 to 1001 and vice versa works, audio is perfect. Packets are propery travelling between 10.99.8.221 and 10.99.8.111
Second case : On windows machine I open gtalk and I open a chat to buddy which is actually a bot logged in on freeswitch (dingaling client mode). The I started java socket program which listens to icoming messages, after typing into client "call 1000@10.99.8.221" an api command is executed: "api originate sofia/default/1000@10.99.8.221 &bridge(dingaling/ gmail.com/gtalk_mail(at)gmail.com<http://gmail.com/gtalk_mail%28at%29gmail.com> )"
A call is placed between gtalk and sip phone 1000, it rings, but when both end answers there is no audio. After a minute, the call ends itself. I've attached wireshark dumps from both ends - what is strange is that packets are not trying to got at right IP, instead they hit some other machine (213.x.x.x), which doesn't make sense.
Fresh log from freeswitch (I don't know why 213.x.x.x gets mixed in this story): http://pastebin.com/m75b10388
// I hope the attachments go trough - 17 KB. test_gtalk_client_side - dump from win machine (gtalk client) test_sip_client - dump from linux machine (freeswitch and sip phone client)
I hope to get resolved this mistery somehow.
Thank you for all kind answers.
_______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
Thanks, commenting ext-rtp fixed my issue. In case of further problems I'll do what you suggested. Thank you again for all help. On Tue, 23 Dec 2008 16:03:38 +0100, Anthony Minessale <anthony.minessale@gmail.com> wrote:
when 2 devices talk via googles gtalk when they are both behind the same lan you are going to have problems.
on thing you can do is make an acl to ignore any candidates that are not local add this to your dingaling profile <param name="candidate-acl" value="myacl"/>
then add myacl to acl.conf.xml that only allows your lan ip.
Turn off all the stun and ext-rtp-ip setting.
OR
use the windows machine from a box that is not on the sam lan behind the same nat.
On Tue, Dec 23, 2008 at 8:09 AM, kriko <kristjan.ugrin@gmail.com> wrote:
I've decided to do this properly: clean fresweetch reinstall.
My worsktation hosts freeswitch + 1 sip phone also running as 1000 (linux - IP 10.99.8.221) Other windows machine has gtalk with and also a sip phone registered as 1001 (IP 10.99.8.111).
First case - SIP to SIP. Calling from 1000 to 1001 and vice versa works, audio is perfect. Packets are propery travelling between 10.99.8.221 and 10.99.8.111
Second case : On windows machine I open gtalk and I open a chat to buddy which is actually a bot logged in on freeswitch (dingaling client mode). The I started java socket program which listens to icoming messages, after typing into client "call 1000@10.99.8.221" an api command is executed: "api originate sofia/default/1000@10.99.8.221 &bridge(dingaling/ gmail.com/gtalk_mail(at)gmail.com<http://gmail.com/gtalk_mail%28at%29gmail.com> )"
A call is placed between gtalk and sip phone 1000, it rings, but when both end answers there is no audio. After a minute, the call ends itself. I've attached wireshark dumps from both ends - what is strange is that packets are not trying to got at right IP, instead they hit some other machine (213.x.x.x), which doesn't make sense.
Fresh log from freeswitch (I don't know why 213.x.x.x gets mixed in this story): http://pastebin.com/m75b10388
// I hope the attachments go trough - 17 KB. test_gtalk_client_side - dump from win machine (gtalk client) test_sip_client - dump from linux machine (freeswitch and sip phone client)
I hope to get resolved this mistery somehow.
Thank you for all kind answers.
_______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Porn - the reason you need a new hard drive.
participants (2)
-
Anthony Minessale
-
kriko