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bypass rtp traffic.

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Kamailio without sound, what to do?

MingHon

13 Jul 2011 13 Jul '11

7:43 a.m.

Hi List, i would like to know is it possible to bypass the rtp traffic forwarding to asterisk server? my kamailio and rtpproxy is on the same box and asterisk is on the other box. can kamailio/rtpproxy handle the rtp traffic without forwarding to asterisk box? thanks in advance. -- Regards, MingHon

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Klaus Darilion

13 Jul 13 Jul

9:56 a.m.

I guess you forward all calls via Asterisk. Yes: set canreinvite=yes (name was changed in newer Asterisk versions) in sip.conf for the peers and Asterisk will send reINVITEs after call setup to offload RTP. regards Klaus Am 13.07.2011 07:43, schrieb MingHon:

Hi List, i would like to know is it possible to bypass the rtp traffic forwarding to asterisk server? my kamailio and rtpproxy is on the same box and asterisk is on the other box. can kamailio/rtpproxy handle the rtp traffic without forwarding to asterisk box? thanks in advance. -- Regards, MingHon _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

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MingHon

10:07 a.m.

Hi, i tried set canreinvite=yes in asterisk but kamailio/rtpproxy still trying to send rtp traffic to asterisk. and asterisk did not forward the rtp traffic back to kamailio/rtpproxy then i will get no audio on the ua. please adv. thanks, Regards, MingHon

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MingHon

14 Jul 14 Jul

4:39 p.m.

Hello, anyone? currently my setup look like this. when UA1 call UA2 and the rtp traffic flow to kamailio and asterisk. [UA1] <--(rtp)--> [Kamailio/RTPProxy] <--(rtp)--> [UA2] ^ | RTP TRAFFIC | v [ASTERISK] what i need to achieve is UA1 call UA2 and rtp traffic flow to kamailio but not to asterisk. can kamailio handle the rtp traffic it own? [UA1] <--(rtp)--> [Kamailio/RTPProxy] <--(rtp)--> [UA2] ^ | X | v [ASTERISK] Thanks in advance. -- Regards, MingHon

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Carsten Bock

5:22 p.m.

Hi, have you tried "canreinvite=yes" on your Asterisk-box? If that does not help, there is probably no way to make the RTP-Traffic bypass your asterisk box... Carsten 2011/7/14 MingHon <gminghon(a)gmail.com>om>:

Hello, anyone? currently my setup look like this. when UA1 call UA2 and the rtp traffic flow to kamailio and asterisk. [UA1] <--(rtp)--> [Kamailio/RTPProxy] <--(rtp)--> [UA2] ^ | RTP TRAFFIC | v [ASTERISK] what i need to achieve is UA1 call UA2 and rtp traffic flow to kamailio but not to asterisk. can kamailio handle the rtp traffic it own? [UA1] <--(rtp)--> [Kamailio/RTPProxy] <--(rtp)--> [UA2] ^ | X | v [ASTERISK] Thanks in advance. -- Regards, MingHon

-- Carsten Bock http://www.ng-voice.com mailto:carsten@ng-voice.com Schomburgstr. 80 22767 Hamburg Germany Mobile +49 179 2021244 Office +49 40 34927219 Fax +49 40 34927220 ~~~~~ Checkout SIP-Provider CE: http://www.sipwise.com/products/spce/overview/

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MingHon

6:09 p.m.

Hi, yup i tried "canreinivte=yes" in sip.conf and also in the extension database. urm how bout having direct rtp traffic and also relay rtp traffic in my setup? example, UA1 and UA2 is at the same nat. so UA1 and UA2 will have direct rtp traffic. UA1 <--(rtp)--> UA2. and UA3 and UA4 both behind different nat will need relay rtp traffic. when invite compare the "received:ip_address". UA3 <--(rtp)--> KAMAILIO <--(rtp)--> UA4 issit possible to have both? urm ya what is the variable for the "received:ip_address" ? Thanks. -- Regards, MingHon On Thu, Jul 14, 2011 at 11:22 PM, Carsten Bock <carsten(a)ng-voice.com> wrote:

Hi, have you tried "canreinvite=yes" on your Asterisk-box? If that does not help, there is probably no way to make the RTP-Traffic bypass your asterisk box... Carsten 2011/7/14 MingHon <gminghon(a)gmail.com>om>:

Hello, anyone? currently my setup look like this. when UA1 call UA2 and the rtp traffic flow to kamailio and asterisk. [UA1] <--(rtp)--> [Kamailio/RTPProxy] <--(rtp)--> [UA2] ^ | RTP TRAFFIC | v [ASTERISK] what i need to achieve is UA1 call UA2 and rtp traffic flow to kamailio

but

not to asterisk. can kamailio handle the rtp traffic it own? [UA1] <--(rtp)--> [Kamailio/RTPProxy] <--(rtp)--> [UA2] ^ | X | v [ASTERISK] Thanks in advance. -- Regards, MingHon

-- Carsten Bock http://www.ng-voice.com mailto:carsten@ng-voice.com Schomburgstr. 80 22767 Hamburg Germany Mobile +49 179 2021244 Office +49 40 34927219 Fax +49 40 34927220 ~~~~~ Checkout SIP-Provider CE: http://www.sipwise.com/products/spce/overview/

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Klaus Darilion

5:40 p.m.

Am 13.07.2011 10:07, schrieb MingHon:

Hi, i tried set canreinvite=yes in asterisk but kamailio/rtpproxy still trying to send rtp traffic to asterisk.

That should not happen. You have to investigate why. You have to take a look at the SIP signaling during and after call setup. You should see reINVITE messages from Asterisk to the clients. Take a look at the SDPs in those requests and their responses to find out if they are malformed. regards Klaus

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MingHon

15 Jul 15 Jul

11:36 a.m.

Hi, after asterisk reinvite i get status 491: request pending. after few seconds i hang up both UA then one of the UA will start ring. please advice. -- Thanks and Regards, MingHon

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Iñaki Baz Castillo

16 Jul 16 Jul

11:03 a.m.

Hi. Kamailio and rtpproxy does NOT decide to send rtp to asterisk. It is asterisk who decides to receive it and that entirely depends on asterisk sip condigurarion and asterisk sip peers configuration. Your question is not related to kamailio but just to asterisk.

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MingHon

21 Jul 21 Jul

9:23 a.m.

Hello List, im still trying but no luck. asterisk canreinvite already set to yes now im testing in lan i setup kamailio and asterisk in same lan kamailio&rtpproxy on 192.168.2.3 and asterisk on 192.168.2.23 canreinvite=yes in asterisk. when both ua in the same lan register directly to asterisk the reinvite work. both ua will have and direct media flow [ua1]<====>[ua2] | | x | v [asterisk] when ua register to kamailio the audio work and the reinvite message is same as the first invite message. [ua1]<====>[kamailio]<====>[ua2] | ^ | | | | v | [asterisk] how do i stop the media flow between kamailio and asterisk? make kamailio relay the rtp between both ua. [ua1]<====>[kamailio]<====>[ua2] | ^ x x | | v | [asterisk] anyone could give some hint? thanks in adv. -- Regards, MingHon

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Klaus Darilion

9:48 a.m.

You should post a SIP trace, together with the IP addresses of all nodes: ngrep -t -d any -P "" -Wbyline port 5060 If there is sensitive information in the traces, just remove/replace it. regards Klaus Am 21.07.2011 09:23, schrieb MingHon:

Hello List, im still trying but no luck. asterisk canreinvite already set to yes now im testing in lan i setup kamailio and asterisk in same lan kamailio&rtpproxy on 192.168.2.3 and asterisk on 192.168.2.23 canreinvite=yes in asterisk. when both ua in the same lan register directly to asterisk the reinvite work. both ua will have and direct media flow [ua1]<====>[ua2] | | x | v [asterisk] when ua register to kamailio the audio work and the reinvite message is same as the first invite message. [ua1]<====>[kamailio]<====>[ua2] | ^ | | | | v | [asterisk] how do i stop the media flow between kamailio and asterisk? make kamailio relay the rtp between both ua. [ua1]<====>[kamailio]<====>[ua2] | ^ x x | | v | [asterisk] anyone could give some hint? thanks in adv. -- Regards, MingHon _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

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Skyler

26 Jul 26 Jul

2:02 p.m.

Hi, On Thu, 2011-07-21 at 15:23 +0800, MingHon wrote:

Hello List, im still trying but no luck. asterisk canreinvite already set to yes

What version of asterisk? I think in 1.6.2 canreinvite was replaced with directmedia and directrtp.

now im testing in lan i setup kamailio and asterisk in same lan kamailio&rtpproxy on 192.168.2.3 and asterisk on 192.168.2.23 canreinvite=yes in asterisk. when both ua in the same lan register directly to asterisk the reinvite work. both ua will have and direct media flow [ua1]<====>[ua2] | | x | v [asterisk] when ua register to kamailio the audio work and the reinvite message is same as the first invite message. [ua1]<====>[kamailio]<====>[ua2] | ^ | | | | v | [asterisk] how do i stop the media flow between kamailio and asterisk? make kamailio relay the rtp between both ua. [ua1]<====>[kamailio]<====>[ua2] | ^ x x | | v | [asterisk] anyone could give some hint? thanks in adv. -- Regards, MingHon

S.

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MingHon

21 Jul 21 Jul

10:31 a.m.

Hi klaus, here is my ngrep i paste it pastebin pls take a look. http://pastebin.com/eHtMXvEx thanks in adv. -- Regards, MingHon

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MingHon

25 Jul 25 Jul

4:37 a.m.

anyone??

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Klaus Darilion

10:51 a.m.

Actually the trace looks fine. You have to debug on network level: check if RTP packets are sent. Your scenario looks like: Caller ---- rtpproxy(a)---rtpproxy(b)----callee. You have 2 instances of rtpproxy activated - which should send RTP packets to each other. e.g. use 'ngrep -d any -t -q -P "" "" udp' to capture all UDP packets and watch if RTP packets are sent by the phones and proper forwarded by rtpproxy. regards Klaus Am 21.07.2011 10:31, schrieb MingHon:

Hi klaus, here is my ngrep i paste it pastebin pls take a look. http://pastebin.com/eHtMXvEx thanks in adv. -- Regards, MingHon

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participants (5)

Carsten Bock
Iñaki Baz Castillo
Klaus Darilion
MingHon
Skyler