Re: [Freeswitch-users] SIP Invite IP fragmentation issue
Ok, my bad. Ethereal for some reason was showing only the first fragment (ethereal bug?). But, now it seems I have hit another problem - it seems that the SIP invites (which are fragmented) are being dropped by the firewall in between us and the SIP provider. Is it possible to shrink the size of the SIP invite so that it fits in a single packet? Any optional stuff in the SIP invite that is sent, that can be thrown away? -Saurabh From: [EMAIL PROTECTED]: [EMAIL PROTECTED]: Tue, 18 Nov 2008 07:34:11 +Subject: [Freeswitch-users] SIP Invite IP fragmentation issue I am having an *odd* issue, which i am not sure freeswitch is to be blamed for. Sometimes, the SIP invites are bigger than 1500 bytes causing IP fragmentation, but when I look at the TCP dump (on the same machine as freeswitch), I see that only the first packet of the fragment is captured. Is freeswitch trying to do its own IP fragmentation or is it relying on underlying linux (kernel 2.6.18)? I created a small program to send UDP packets of 2000 bytes, and also tried with ping -s 2000, and both were successful, so am leaning towards blaming Freeswitch. Any suggestions? -Saurabh Stay up to date on your PC, the Web, and your mobile phone with Windows Live Click here _ Stay up to date on your PC, the Web, and your mobile phone with Windows Live http://clk.atdmt.com/MRT/go/119462413/direct/01/___ 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
Re: [Freeswitch-users] SIP Invite IP fragmentation issue
enabling compact headers - what is that? -Saurabh Date: Tue, 18 Nov 2008 04:29:28 -0600From: [EMAIL PROTECTED]: [EMAIL PROTECTED]: Re: [Freeswitch-users] SIP Invite IP fragmentation issueIts not really possible other then enabling compact headers or by getting rid of codecs that you don’t actually want to use... Another thing you could do is get your broken ISP to fix their firewall... It is not correct to just drop fragmented packets just because they are fragmented.. This is something that will happen on a regular basis on the internet as not everything has an MTU of 1500 From: Saurabh Aggarwal [EMAIL PROTECTED]Reply-To: freeswitch-users@lists.freeswitch.orgDate: Tue, 18 Nov 2008 10:19:55 +To: freeswitch-users@lists.freeswitch.orgSubject: Re: [Freeswitch-users] SIP Invite IP fragmentation issueOk, my bad. Ethereal for some reason was showing only the first fragment (ethereal bug?). But, now it seems I have hit another problem - it seems that the SIP invites (which are fragmented) are being dropped by the firewall in between us and the SIP provider. Is it possible to shrink the size of the SIP invite so that it fits in a single packet? Any optional stuff in the SIP invite that is sent, that can be thrown away? -Saurabh From: [EMAIL PROTECTED]: [EMAIL PROTECTED]: Tue, 18 Nov 2008 07:34:11 +Subject: [Freeswitch-users] SIP Invite IP fragmentation issueI am having an *odd* issue, which i am not sure freeswitch is to be blamed for. Sometimes, the SIP invites are bigger than 1500 bytes causing IP fragmentation, but when I look at the TCP dump (on the same machine as freeswitch), I see that only the first packet of the fragment is captured. Is freeswitch trying to do its own IP fragmentation or is it relying on underlying linux (kernel 2.6.18)? I created a small program to send UDP packets of 2000 bytes, and also tried with ping -s 2000, and both were successful, so am leaning towards blaming Freeswitch. Any suggestions? -Saurabh Stay up to date on your PC, the Web, and your mobile phone with Windows Live Click here http://clk.atdmt.com/MRT/go/119462413/direct/01/ Stay up to date on your PC, the Web, and your mobile phone with Windows Live Click here http://clk.atdmt.com/MRT/go/119462413/direct/01/ ___Freeswitch-users mailing [EMAIL PROTECTED]://lists.freeswitch.org/mailman/listinfo/freeswitch-usersUNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org _ Get more done, have more fun, and stay more connected with Windows Mobile®. http://clk.atdmt.com/MRT/go/119642556/direct/01/___ 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
Re: [Freeswitch-users] SIP Invite IP fragmentation issue
Thanks, how do I enable this in freeswitch? Can this be done through the SIP configuration file? -Saurabh Date: Tue, 18 Nov 2008 12:05:18 +0100From: [EMAIL PROTECTED]: [EMAIL PROTECTED]: Re: [Freeswitch-users] SIP Invite IP fragmentation issueThe rfc also describes why:SIP provides a mechanism to represent common header field names in an abbreviated form. This may be useful when messages would otherwise become too large to be carried on the transport available to it (exceeding the maximum transmission unit (MTU) when using UDP, for example). These compact forms are defined in Section 20. A compact form MAY be substituted for the longer form of a header field name at any time without changing the semantics of the message. A header field name MAY appear in both long and short forms within the same message. Implementations MUST accept both the long and short forms of each header name. On Tue, Nov 18, 2008 at 11:52 AM, Iñaki Baz Castillo [EMAIL PROTECTED] wrote: 2008/11/18 Saurabh Aggarwal [EMAIL PROTECTED]: enabling compact headers - what is that?SIP allows compact headers names for a few heades: From = f To = t Via = v ...--Iñaki Baz Castillo[EMAIL PROTECTED]___Freeswitch-users mailing [EMAIL PROTECTED]://lists.freeswitch.org/mailman/listinfo/freeswitch-usersUNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org _ Get more done, have more fun, and stay more connected with Windows Mobile®. http://clk.atdmt.com/MRT/go/119642556/direct/01/___ 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
[Freeswitch-users] SIP Invite IP fragmentation issue
I am having an *odd* issue, which i am not sure freeswitch is to be blamed for. Sometimes, the SIP invites are bigger than 1500 bytes causing IP fragmentation, but when I look at the TCP dump (on the same machine as freeswitch), I see that only the first packet of the fragment is captured. Is freeswitch trying to do its own IP fragmentation or is it relying on underlying linux (kernel 2.6.18)? I created a small program to send UDP packets of 2000 bytes, and also tried with ping -s 2000, and both were successful, so am leaning towards blaming Freeswitch. Any suggestions? -Saurabh _ Stay up to date on your PC, the Web, and your mobile phone with Windows Live http://clk.atdmt.com/MRT/go/119462413/direct/01/___ 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
Re: [Freeswitch-users] SIP incoming call routing
Thanks - that does work to an extent. Now the problem is that not all gateways would allow arbitrary extensions. E.g. AIM Callout - it *requires* that the extension/caller-id be your aim username. -Saurabh Date: Wed, 29 Oct 2008 12:46:44 -0500From: [EMAIL PROTECTED]: [EMAIL PROTECTED]: Re: [Freeswitch-users] SIP incoming call routingwhatever you put in the extension param in the gateway should control what destination_number it has in the inbound call. you can also do your regex in your dialplan on any of the info in the sip packet besides destination number if you wish. On Wed, Oct 29, 2008 at 4:52 AM, Saurabh Aggarwal [EMAIL PROTECTED] wrote: Yes, but there is no DID in my system for incoming calls. I have users dynamically registering gateways, and calls coming in to SIP ids that they have used to register. -Saurabh Date: Wed, 29 Oct 2008 15:12:28 +0530From: [EMAIL PROTECTED]: [EMAIL PROTECTED]: Re: [Freeswitch-users] SIP incoming call routing On Wed, Oct 29, 2008 at 2:52 PM, Saurabh Aggarwal [EMAIL PROTECTED] wrote: We are using freeswitch as a SIP proxy, where we are letting people register with freeswitch, and in-turn we do the SIP registration for them to arbitrary sip servers (as requested by users) - each user gets his own sip gateway in the freeswitch configuration. Then they can make outgoing calls and calls are routed through their specific SIP gateway. Now the problem is that when a call is received from one of these SIP registrations, it hits the public.xml where I can't seem to figure out how to get the SIP gateway information from which it came in. The SIP gateway name actually contains the information where it should be routed to. Any ideas on how to approach this problem? Question - is it possible to do it in the dialplan (dynamic) or do we have to write an application to do this mapping? -Saurabh have you looked at this example http://wiki.freeswitch.org/wiki/Getting_Started_Guide#Receiving_an_inbound_call_from_a_Gateway ram When your life is on the go—take your life with you. Try Windows Mobile® today___Freeswitch-users mailing [EMAIL PROTECTED]://lists.freeswitch.org/mailman/listinfo/freeswitch-usersUNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org-- Anthony Minessale IIFreeSWITCH http://www.freeswitch.org/ClueCon http://www.cluecon.com/AIM: anthmMSN:[EMAIL PROTECTED]/JABBER/PAYPAL:[EMAIL PROTECTED]: irc.freenode.net #freeswitchFreeSWITCH Developer Conferencesip:[EMAIL PROTECTED]:[EMAIL PROTECTED]/888googletalk:[EMAIL PROTECTED]:213-799-1400 _ When your life is on the go—take your life with you. http://clk.atdmt.com/MRT/go/115298558/direct/01/___ 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
[Freeswitch-users] RTMP Support (Flash)
Any plans on adding RTMP support to freeswitch - that could be a real killer feature, allowing flash clients to call into Freeswitch. Now that Red5 has done all the hard work, it would be pretty cool if an endpoint can be developed. -Saurabh _ Want to read Hotmail messages in Outlook? The Wordsmiths show you how. http://windowslive.com/connect/post/wedowindowslive.spaces.live.com-Blog-cns!20EE04FBC541789!167.entry?ocid=TXT_TAGLM_WL_hotmail_092008___ 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
Re: [Freeswitch-users] RTMP Support (Flash)
There's another project (porting Red5 to C++) - http://code.google.com/p/red5cpp/ -Saurabh From: [EMAIL PROTECTED]: [EMAIL PROTECTED]: Tue, 4 Nov 2008 07:18:54 +Subject: Re: [Freeswitch-users] RTMP Support (Flash) Yup. But, still would be a killer feature if it can be done. Click to Call from web is all freeswitch needs to differentiate itself from Asterisk. The Jingle endpoint (first at freeswitch) was something that got me over to freeswitch, I am sure RTMP would get a lot more. -Saurabh Date: Mon, 3 Nov 2008 16:01:10 -0600From: [EMAIL PROTECTED]: [EMAIL PROTECTED]: Re: [Freeswitch-users] RTMP Support (Flash)doh, it's in java! That might be a little more challenging. On Mon, Nov 3, 2008 at 3:12 PM, Anthony Minessale [EMAIL PROTECTED] wrote: it's licensed lgpl, so it's compatible so sure why not! On Mon, Nov 3, 2008 at 6:55 AM, Saurabh Aggarwal [EMAIL PROTECTED] wrote: Any plans on adding RTMP support to freeswitch - that could be a real killer feature, allowing flash clients to call into Freeswitch. Now that Red5 has done all the hard work, it would be pretty cool if an endpoint can be developed. -Saurabh Want to read Hotmail messages in Outlook? The Wordsmiths show you how. Learn Now___Freeswitch-users mailing [EMAIL PROTECTED]://lists.freeswitch.org/mailman/listinfo/freeswitch-usersUNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org-- Anthony Minessale IIFreeSWITCH http://www.freeswitch.org/ClueCon http://www.cluecon.com/AIM: anthmMSN:[EMAIL PROTECTED]/JABBER/PAYPAL:[EMAIL PROTECTED]: irc.freenode.net #freeswitchFreeSWITCH Developer Conferencesip:[EMAIL PROTECTED]:[EMAIL PROTECTED]/888googletalk:[EMAIL PROTECTED]:213-799-1400-- Anthony Minessale IIFreeSWITCH http://www.freeswitch.org/ClueCon http://www.cluecon.com/AIM: anthmMSN:[EMAIL PROTECTED]/JABBER/PAYPAL:[EMAIL PROTECTED]: irc.freenode.net #freeswitchFreeSWITCH Developer Conferencesip:[EMAIL PROTECTED]:[EMAIL PROTECTED]/888googletalk:[EMAIL PROTECTED]:213-799-1400 Store, manage and share up to 5GB with Windows Live SkyDrive. Start uploading now _ Stay organized with simple drag and drop from Windows Live Hotmail. http://windowslive.com/Explore/hotmail?ocid=TXT_TAGLM_WL_hotmail_102008___ 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
Re: [Freeswitch-users] RTMP Support (Flash)
Yup. But, still would be a killer feature if it can be done. Click to Call from web is all freeswitch needs to differentiate itself from Asterisk. The Jingle endpoint (first at freeswitch) was something that got me over to freeswitch, I am sure RTMP would get a lot more. -Saurabh Date: Mon, 3 Nov 2008 16:01:10 -0600From: [EMAIL PROTECTED]: [EMAIL PROTECTED]: Re: [Freeswitch-users] RTMP Support (Flash)doh, it's in java! That might be a little more challenging. On Mon, Nov 3, 2008 at 3:12 PM, Anthony Minessale [EMAIL PROTECTED] wrote: it's licensed lgpl, so it's compatible so sure why not! On Mon, Nov 3, 2008 at 6:55 AM, Saurabh Aggarwal [EMAIL PROTECTED] wrote: Any plans on adding RTMP support to freeswitch - that could be a real killer feature, allowing flash clients to call into Freeswitch. Now that Red5 has done all the hard work, it would be pretty cool if an endpoint can be developed. -Saurabh Want to read Hotmail messages in Outlook? The Wordsmiths show you how. Learn Now___Freeswitch-users mailing [EMAIL PROTECTED]://lists.freeswitch.org/mailman/listinfo/freeswitch-usersUNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org-- Anthony Minessale IIFreeSWITCH http://www.freeswitch.org/ClueCon http://www.cluecon.com/AIM: anthmMSN:[EMAIL PROTECTED]/JABBER/PAYPAL:[EMAIL PROTECTED]: irc.freenode.net #freeswitchFreeSWITCH Developer Conferencesip:[EMAIL PROTECTED]:[EMAIL PROTECTED]/888googletalk:[EMAIL PROTECTED]:213-799-1400-- Anthony Minessale IIFreeSWITCH http://www.freeswitch.org/ClueCon http://www.cluecon.com/AIM: anthmMSN:[EMAIL PROTECTED]/JABBER/PAYPAL:[EMAIL PROTECTED]: irc.freenode.net #freeswitchFreeSWITCH Developer Conferencesip:[EMAIL PROTECTED]:[EMAIL PROTECTED]/888googletalk:[EMAIL PROTECTED]:213-799-1400 _ Store, manage and share up to 5GB with Windows Live SkyDrive. http://skydrive.live.com/welcome.aspx?provision=1?ocid=TXT_TAGLM_WL_skydrive_102008___ 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
Re: [Freeswitch-users] SIP incoming call routing
Yes, but there is no DID in my system for incoming calls. I have users dynamically registering gateways, and calls coming in to SIP ids that they have used to register. -Saurabh Date: Wed, 29 Oct 2008 15:12:28 +0530From: [EMAIL PROTECTED]: [EMAIL PROTECTED]: Re: [Freeswitch-users] SIP incoming call routing On Wed, Oct 29, 2008 at 2:52 PM, Saurabh Aggarwal [EMAIL PROTECTED] wrote: We are using freeswitch as a SIP proxy, where we are letting people register with freeswitch, and in-turn we do the SIP registration for them to arbitrary sip servers (as requested by users) - each user gets his own sip gateway in the freeswitch configuration. Then they can make outgoing calls and calls are routed through their specific SIP gateway. Now the problem is that when a call is received from one of these SIP registrations, it hits the public.xml where I can't seem to figure out how to get the SIP gateway information from which it came in. The SIP gateway name actually contains the information where it should be routed to. Any ideas on how to approach this problem? Question - is it possible to do it in the dialplan (dynamic) or do we have to write an application to do this mapping? -Saurabh have you looked at this example http://wiki.freeswitch.org/wiki/Getting_Started_Guide#Receiving_an_inbound_call_from_a_Gateway ram _ When your life is on the go—take your life with you. http://clk.atdmt.com/MRT/go/115298558/direct/01/___ 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
[Freeswitch-users] SIP incoming call routing
We are using freeswitch as a SIP proxy, where we are letting people register with freeswitch, and in-turn we do the SIP registration for them to arbitrary sip servers (as requested by users) - each user gets his own sip gateway in the freeswitch configuration. Then they can make outgoing calls and calls are routed through their specific SIP gateway. Now the problem is that when a call is received from one of these SIP registrations, it hits the public.xml where I can't seem to figure out how to get the SIP gateway information from which it came in. The SIP gateway name actually contains the information where it should be routed to. Any ideas on how to approach this problem? Question - is it possible to do it in the dialplan (dynamic) or do we have to write an application to do this mapping? -Saurabh _ When your life is on the go—take your life with you. http://clk.atdmt.com/MRT/go/115298558/direct/01/___ 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
