Hi all, I compiled FreeSWITCH Version 1.0.5-20100209-0400 (16587M) with mod_h323. When I call from h323 (Avaya IPOffice) to local fs extention (x-lite) I hear ring back, but when x-lite picks up, he hears silence, while IPOffice user continues to hear ringback. The log is at the http://pastebin.freeswitch.org/12091 My configuration is almost default, several local extentions added, and h323.conf from http://wiki.freeswitch.org/wiki/Mod_h323 5840 - user at IPOffice 2853 - x-lite registered at FS IPOffice ip address: 172.23.14.2 FS ip address 172.23.22.49 Can anybody please advise how to solve that? Is it a configuration or a software problem? Thanks in advance, Nikolay.
I forgot to add, that from tcpdump trace (and from log) one can see that FS does not send answer to IPOffice. _____ From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Nikolay Kondratyev Sent: Wednesday, February 10, 2010 1:49 PM To: freeswitch-users@lists.freeswitch.org Subject: [Freeswitch-users] h323 - sip call is not working Hi all, I compiled FreeSWITCH Version 1.0.5-20100209-0400 (16587M) with mod_h323. When I call from h323 (Avaya IPOffice) to local fs extention (x-lite) I hear ring back, but when x-lite picks up, he hears silence, while IPOffice user continues to hear ringback. The log is at the http://pastebin.freeswitch.org/12091 My configuration is almost default, several local extentions added, and h323.conf from http://wiki.freeswitch.org/wiki/Mod_h323 5840 - user at IPOffice 2853 - x-lite registered at FS IPOffice ip address: 172.23.14.2 FS ip address 172.23.22.49 Can anybody please advise how to solve that? Is it a configuration or a software problem? Thanks in advance, Nikolay.
On Wed, Feb 10, 2010 at 11:49 AM, Nikolay Kondratyev <kond@nstel.ru> wrote:
Hi all,
I compiled FreeSWITCH Version 1.0.5-20100209-0400 (16587M) with mod_h323.
When I call from h323 (Avaya IPOffice) to local fs extention (x-lite) I hear ring back, but when x-lite picks up, he hears silence, while IPOffice user continues to hear ringback.
The log is at the http://pastebin.freeswitch.org/12091
My configuration is almost default, several local extentions added, and h323.conf from http://wiki.freeswitch.org/wiki/Mod_h323
5840 – user at IPOffice
2853 – x-lite registered at FS
IPOffice ip address: 172.23.14.2
FS ip address 172.23.22.49
Can anybody please advise how to solve that?
Is it a configuration or a software problem?
Please can you send me the tcpdump as well (not filtered). IPOffice i known to have a "broken" H323 stack. did you try to play with tunneling and h245 in setup settings as well ? It looks like there is some h245 negotiation still pending but cant see that from the logs.
Thanks in advance,
Nikolay.
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I've been running both h323 and SIP between FS and Avaya IPO for some time. No problems at all. :) But make sure to enable h323 fast start and disable "direct media path" in the IPO, if I remember correctly these where the only two parameters that made any real difference for me. But I do recommenf to use SIP, since it's much better supported by FS. /Peter ________________________________________ Från: freeswitch-users-bounces@lists.freeswitch.org [freeswitch-users-bounces@lists.freeswitch.org] för Tihomir Culjaga [tculjaga@gmail.com] Skickat: den 10 februari 2010 20:13 Till: freeswitch-users@lists.freeswitch.org Ämne: Re: [Freeswitch-users] h323 - sip call is not working On Wed, Feb 10, 2010 at 11:49 AM, Nikolay Kondratyev <kond@nstel.ru<mailto:kond@nstel.ru>> wrote: Hi all, I compiled FreeSWITCH Version 1.0.5-20100209-0400 (16587M) with mod_h323. When I call from h323 (Avaya IPOffice) to local fs extention (x-lite) I hear ring back, but when x-lite picks up, he hears silence, while IPOffice user continues to hear ringback. The log is at the http://pastebin.freeswitch.org/12091 My configuration is almost default, several local extentions added, and h323.conf from http://wiki.freeswitch.org/wiki/Mod_h323 5840 – user at IPOffice 2853 – x-lite registered at FS IPOffice ip address: 172.23.14.2 FS ip address 172.23.22.49 Can anybody please advise how to solve that? Is it a configuration or a software problem? Please can you send me the tcpdump as well (not filtered). IPOffice i known to have a "broken" H323 stack. did you try to play with tunneling and h245 in setup settings as well ? It looks like there is some h245 negotiation still pending but cant see that from the logs. Thanks in advance, Nikolay. _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org<mailto: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 !DSPAM:4b73075e32931909716500!
On Wed, Feb 10, 2010 at 8:42 PM, Peter Olsson < peter.olsson@visionutveckling.se> wrote:
I've been running both h323 and SIP between FS and Avaya IPO for some time. No problems at all. :)
But make sure to enable h323 fast start and disable "direct media path" in the IPO, if I remember correctly these where the only two parameters that made any real difference for me.
But I do recommenf to use SIP, since it's much better supported by FS.
/Peter
Peter, what H323plus version are you using ? did you noticed q931 release cause is not mapped H323 => SIP correctly ?
I'm using the opal module, since mod_h323 is not available in Windows yet. We're not so dependant of all the correct release codes etc, we use it mostly for ASR and voicemail applications, but I do know there have been discussions about this on the mailing list. My recommendation is to use SIP to the IPO instead (mostly beacuse SIP is much better supported by FS), we're using that more and more. /Peter Från: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] För Tihomir Culjaga Skickat: den 10 februari 2010 21:27 Till: freeswitch-users@lists.freeswitch.org Ämne: Re: [Freeswitch-users] h323 - sip call is not working On Wed, Feb 10, 2010 at 8:42 PM, Peter Olsson <peter.olsson@visionutveckling.se<mailto:peter.olsson@visionutveckling.se>> wrote: I've been running both h323 and SIP between FS and Avaya IPO for some time. No problems at all. :) But make sure to enable h323 fast start and disable "direct media path" in the IPO, if I remember correctly these where the only two parameters that made any real difference for me. But I do recommenf to use SIP, since it's much better supported by FS. /Peter Peter, what H323plus version are you using ? did you noticed q931 release cause is not mapped H323 => SIP correctly ? !DSPAM:4b73196232931389031035!
Nikolay, you are sending slow start with tunneling=true ?!?! It is not gong to work :) Please can you set fast start instead? Your call failed because there was no mediaControll channel negotiated at all... actually the call had to be aborted because wrong signaling .. but anyhow. Please on your IPO use FastStart with h245Tunneling=true... also, same settings on FS side as well (exclude h245 in setup as well). Frame 13 (277 bytes on wire, 277 bytes captured) Ethernet II, Src: AlliedTe_22:9b:4a (00:00:cd:22:9b:4a), Dst: Vmware_67:33:a7 (00:0c:29:67:33:a7) Internet Protocol, Src: 172.23.14.2 (172.23.14.2), Dst: 172.23.22.49 (172.23.22.49) Transmission Control Protocol, Src Port: oirtgsvc (4141), Dst Port: h323hostcall (1720), Seq: 1, Ack: 1, Len: 223 TPKT, Version: 3, Length: 223 Q.931 Protocol discriminator: Q.931 Call reference value length: 2 Call reference flag: Message sent from originating side Call reference value: 0012 Message type: SETUP (0x05) Bearer capability Display 'Gornak Alexandr>2853' Calling party number: '5840' Called party number: '2853' User-user H.225.0 CS H323-UserInformation h323-uu-pdu h323-message-body: setup (0) setup h4501SupplementaryService: 1 item * 1... .... h245Tunneling: True* On Thu, Feb 11, 2010 at 9:18 AM, Nikolay Kondratyev <kond@nstel.ru> wrote:
But I do recommenf to use SIP, since it's much better supported by FS.
/Peter But SIP is poorly supported by IPO. Thanks and regards, Nikolay.
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Tihomir, Thanks for help. I enabled fast start on IPO and I can hear voice now. But the ringback tone and voice appears to be wheezy, but I will investigate that tomorrow. Thanks again. Nikolay. _____ From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Tihomir Culjaga Sent: Thursday, February 11, 2010 2:34 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] h323 - sip call is not working Nikolay, you are sending slow start with tunneling=true ?!?! It is not gong to work :) Please can you set fast start instead? Your call failed because there was no mediaControll channel negotiated at all... actually the call had to be aborted because wrong signaling .. but anyhow. Please on your IPO use FastStart with h245Tunneling=true... also, same settings on FS side as well (exclude h245 in setup as well). Frame 13 (277 bytes on wire, 277 bytes captured) Ethernet II, Src: AlliedTe_22:9b:4a (00:00:cd:22:9b:4a), Dst: Vmware_67:33:a7 (00:0c:29:67:33:a7) Internet Protocol, Src: 172.23.14.2 (172.23.14.2), Dst: 172.23.22.49 (172.23.22.49) Transmission Control Protocol, Src Port: oirtgsvc (4141), Dst Port: h323hostcall (1720), Seq: 1, Ack: 1, Len: 223 TPKT, Version: 3, Length: 223 Q.931 Protocol discriminator: Q.931 Call reference value length: 2 Call reference flag: Message sent from originating side Call reference value: 0012 Message type: SETUP (0x05) Bearer capability Display 'Gornak Alexandr>2853' Calling party number: '5840' Called party number: '2853' User-user H.225.0 CS H323-UserInformation h323-uu-pdu h323-message-body: setup (0) setup h4501SupplementaryService: 1 item 1... .... h245Tunneling: True On Thu, Feb 11, 2010 at 9:18 AM, Nikolay Kondratyev <kond@nstel.ru> wrote:
But I do recommenf to use SIP, since it's much better supported by FS.
/Peter
But SIP is poorly supported by IPO. Thanks and regards, Nikolay. _______________________________________________ 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
On Thu, Feb 11, 2010 at 4:18 PM, Nikolay Kondratyev <kond@nstel.ru> wrote:
Tihomir,
Thanks for help.
I enabled fast start on IPO and I can hear voice now. But the ringback tone and voice appears to be wheezy, but I will investigate that tomorrow.
set framing time for your codec to 30ms in IPO, also play with PI in alerting... set it to 2.
Thanks again.
Nikolay.
------------------------------
*From:* freeswitch-users-bounces@lists.freeswitch.org [mailto: freeswitch-users-bounces@lists.freeswitch.org] *On Behalf Of *Tihomir Culjaga *Sent:* Thursday, February 11, 2010 2:34 PM
*To:* freeswitch-users@lists.freeswitch.org *Subject:* Re: [Freeswitch-users] h323 - sip call is not working
Nikolay, you are sending slow start with tunneling=true ?!?!
It is not gong to work :)
Please can you set fast start instead?
Your call failed because there was no mediaControll channel negotiated at all... actually the call had to be aborted because wrong signaling .. but anyhow.
Please on your IPO use FastStart with h245Tunneling=true... also, same settings on FS side as well (exclude h245 in setup as well).
Frame 13 (277 bytes on wire, 277 bytes captured) Ethernet II, Src: AlliedTe_22:9b:4a (00:00:cd:22:9b:4a), Dst: Vmware_67:33:a7 (00:0c:29:67:33:a7) Internet Protocol, Src: 172.23.14.2 (172.23.14.2), Dst: 172.23.22.49 (172.23.22.49) Transmission Control Protocol, Src Port: oirtgsvc (4141), Dst Port: h323hostcall (1720), Seq: 1, Ack: 1, Len: 223 TPKT, Version: 3, Length: 223 Q.931 Protocol discriminator: Q.931 Call reference value length: 2 Call reference flag: Message sent from originating side Call reference value: 0012 Message type: SETUP (0x05) Bearer capability Display 'Gornak Alexandr>2853' Calling party number: '5840' Called party number: '2853' User-user H.225.0 CS H323-UserInformation h323-uu-pdu h323-message-body: setup (0) setup h4501SupplementaryService: 1 item * 1... .... h245Tunneling: True*
On Thu, Feb 11, 2010 at 9:18 AM, Nikolay Kondratyev <kond@nstel.ru> wrote:
But I do recommenf to use SIP, since it's much better supported by FS.
/Peter
But SIP is poorly supported by IPO. Thanks and regards,
Nikolay.
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Tihomir, Thanks for the reply. I was quite surprised to see 30ms packetization time. and first thought it's a typo. But now I see (if I'm not mistaken) that FS really sends open logical channel with Alaw:30ms when opening logical channel during fast start (see attached file). This parameter is "display only" on my IPO. Is it possible to change it on the FS side? I think it should be somewhere in mod_h323 configuration. But I did not found any mod_h323 configuration parameters except ip addr. and port. By the way, I know that one can use different packetization times for the same codec, but I've never heard, that somebody really uses 30 ms for G711Alaw. Always 20ms. Thanks ans regards, Nikolay. _____ From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Tihomir Culjaga Sent: Thursday, February 11, 2010 9:14 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] h323 - sip call is not working On Thu, Feb 11, 2010 at 4:18 PM, Nikolay Kondratyev <kond@nstel.ru> wrote: Tihomir, Thanks for help. I enabled fast start on IPO and I can hear voice now. But the ringback tone and voice appears to be wheezy, but I will investigate that tomorrow. set framing time for your codec to 30ms in IPO, also play with PI in alerting... set it to 2. Thanks again. Nikolay. _____ From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Tihomir Culjaga Sent: Thursday, February 11, 2010 2:34 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] h323 - sip call is not working Nikolay, you are sending slow start with tunneling=true ?!?! It is not gong to work :) Please can you set fast start instead? Your call failed because there was no mediaControll channel negotiated at all... actually the call had to be aborted because wrong signaling .. but anyhow. Please on your IPO use FastStart with h245Tunneling=true... also, same settings on FS side as well (exclude h245 in setup as well). Frame 13 (277 bytes on wire, 277 bytes captured) Ethernet II, Src: AlliedTe_22:9b:4a (00:00:cd:22:9b:4a), Dst: Vmware_67:33:a7 (00:0c:29:67:33:a7) Internet Protocol, Src: 172.23.14.2 (172.23.14.2), Dst: 172.23.22.49 (172.23.22.49) Transmission Control Protocol, Src Port: oirtgsvc (4141), Dst Port: h323hostcall (1720), Seq: 1, Ack: 1, Len: 223 TPKT, Version: 3, Length: 223 Q.931 Protocol discriminator: Q.931 Call reference value length: 2 Call reference flag: Message sent from originating side Call reference value: 0012 Message type: SETUP (0x05) Bearer capability Display 'Gornak Alexandr>2853' Calling party number: '5840' Called party number: '2853' User-user H.225.0 CS H323-UserInformation h323-uu-pdu h323-message-body: setup (0) setup h4501SupplementaryService: 1 item 1... .... h245Tunneling: True On Thu, Feb 11, 2010 at 9:18 AM, Nikolay Kondratyev <kond@nstel.ru> wrote:
But I do recommenf to use SIP, since it's much better supported by FS.
/Peter
But SIP is poorly supported by IPO. Thanks and regards, Nikolay. _______________________________________________ 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 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
On 2010-02-12 14:24 +0300, Nikolay Kondratyev wrote freeswitch-users@lists....: NK>Tihomir, NK> NK>Thanks for the reply. NK> NK> NK> NK>I was quite surprised to see 30ms packetization time. and first thought it's NK>a typo. NK> NK>But now I see (if I'm not mistaken) that FS really sends open logical NK>channel with Alaw:30ms when opening logical channel during fast start (see NK>attached file). NK> NK>This parameter is "display only" on my IPO. NK> NK>Is it possible to change it on the FS side? I think it should be somewhere NK>in mod_h323 configuration. But I did not found any mod_h323 configuration NK>parameters except ip addr. and port. It's not implemented at this time. NK>By the way, I know that one can use different packetization times for the NK>same codec, but I've never heard, that somebody really uses 30 ms for NK>G711Alaw. Always 20ms. NK> NK> NK> NK>Thanks ans regards, NK> NK>Nikolay. NK> NK> NK> NK> NK> NK> NK> NK> _____ NK> NK>From: freeswitch-users-bounces@lists.freeswitch.org NK>[mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Tihomir NK>Culjaga NK>Sent: Thursday, February 11, 2010 9:14 PM NK>To: freeswitch-users@lists.freeswitch.org NK>Subject: Re: [Freeswitch-users] h323 - sip call is not working NK> NK> NK> NK> NK> NK>On Thu, Feb 11, 2010 at 4:18 PM, Nikolay Kondratyev <kond@nstel.ru> wrote: NK> NK>Tihomir, NK> NK>Thanks for help. NK> NK>I enabled fast start on IPO and I can hear voice now. But the ringback tone NK>and voice appears to be wheezy, but I will investigate that tomorrow. NK> NK> NK>set framing time for your codec to 30ms in IPO, also play with PI in NK>alerting... set it to 2. NK> NK> NK> NK>Thanks again. NK> NK>Nikolay. NK> NK> NK> NK> NK> NK> NK> NK> NK> _____ NK> NK> NK>From: freeswitch-users-bounces@lists.freeswitch.org NK>[mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Tihomir NK>Culjaga NK>Sent: Thursday, February 11, 2010 2:34 PM NK> NK> NK>To: freeswitch-users@lists.freeswitch.org NK>Subject: Re: [Freeswitch-users] h323 - sip call is not working NK> NK> NK> NK>Nikolay, you are sending slow start with tunneling=true ?!?! NK> NK> NK> NK>It is not gong to work :) NK> NK>Please can you set fast start instead? NK> NK> NK>Your call failed because there was no mediaControll channel negotiated at NK>all... actually the call had to be aborted because wrong signaling .. but NK>anyhow. NK> NK>Please on your IPO use FastStart with h245Tunneling=true... also, same NK>settings on FS side as well (exclude h245 in setup as well). NK> NK> NK> NK> NK>Frame 13 (277 bytes on wire, 277 bytes captured) NK>Ethernet II, Src: AlliedTe_22:9b:4a (00:00:cd:22:9b:4a), Dst: NK>Vmware_67:33:a7 (00:0c:29:67:33:a7) NK>Internet Protocol, Src: 172.23.14.2 (172.23.14.2), Dst: 172.23.22.49 NK>(172.23.22.49) NK>Transmission Control Protocol, Src Port: oirtgsvc (4141), Dst Port: NK>h323hostcall (1720), Seq: 1, Ack: 1, Len: 223 NK>TPKT, Version: 3, Length: 223 NK>Q.931 NK> Protocol discriminator: Q.931 NK> Call reference value length: 2 NK> Call reference flag: Message sent from originating side NK> Call reference value: 0012 NK> Message type: SETUP (0x05) NK> Bearer capability NK> Display 'Gornak Alexandr>2853' NK> Calling party number: '5840' NK> Called party number: '2853' NK> User-user NK>H.225.0 CS NK> H323-UserInformation NK> h323-uu-pdu NK> h323-message-body: setup (0) NK> setup NK> h4501SupplementaryService: 1 item NK> 1... .... h245Tunneling: True NK> NK> NK> NK> NK> NK> NK> NK>On Thu, Feb 11, 2010 at 9:18 AM, Nikolay Kondratyev <kond@nstel.ru> wrote: NK> NK>> But I do recommenf to use SIP, since it's much better supported by FS. NK>> NK>> /Peter NK> NK>But SIP is poorly supported by IPO. NK>Thanks and regards, NK> NK>Nikolay. NK> NK> NK>_______________________________________________ NK>FreeSWITCH-users mailing list NK>FreeSWITCH-users@lists.freeswitch.org NK> NK>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users NK>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users NK>http://www.freeswitch.org NK> NK> NK> NK> NK>_______________________________________________ NK>FreeSWITCH-users mailing list NK>FreeSWITCH-users@lists.freeswitch.org NK>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users NK>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users NK>http://www.freeswitch.org NK> NK> NK> NK> C уважением With Best Regards Георгиевский Юрий. Georgiewskiy Yuriy +7 4872 711666 +7 4872 711666 факс +7 4872 711143 fax +7 4872 711143 Компания ООО "Ай Ти Сервис" IT Service Ltd http://nkoort.ru http://nkoort.ru JID: GHhost@jabber.tula-ix.net.ru JID: GHhost@jabber.tula-ix.net.ru YG129-RIPE YG129-RIPE
.
NK> NK>This parameter is "display only" on my IPO. NK> NK>Is it possible to change it on the FS side? I think it should be somewhere
It is deep in H323plus... The problem is that IPO should stick to the protocol and adjust the framing size accordingly... Anyhow, yes this is one of the issues that makes trigger a different call flow on the remote side. In my scenario, when cisco PGW receives a setup with different framing size, it triggers a different behaviour.... PGW starts to negotiate framing using Facility OLC. On Avaya S8700 the behavior is clean ... it accepts it. Anyhow, i'm looking for a way to make this config available in the h323.conf.xml T.
NK>in mod_h323 configuration. But I did not found any mod_h323 configuration NK>parameters except ip addr. and port.
It's not implemented at this time.
NK>By the way, I know that one can use different packetization times for the NK>same codec, but I've never heard, that somebody really uses 30 ms for
This is a rather broad assumption. I have seen 40ms, 60ms and even 80ms in the wild. It all depends on what you want to do. It lowers overhead and increases efficiency on the wire. /b On Feb 12, 2010, at 5:24 AM, Nikolay Kondratyev wrote:
By the way, I know that one can use different packetization times for the same codec, but I’ve never heard, that somebody really uses 30 ms for G711Alaw. Always 20ms.
On Fri, Feb 12, 2010 at 3:29 PM, Brian West <brian@freeswitch.org> wrote:
This is a rather broad assumption. I have seen 40ms, 60ms and even 80ms in the wild. It all depends on what you want to do. It lowers overhead and increases efficiency on the wire.
/b
On Feb 12, 2010, at 5:24 AM, Nikolay Kondratyev wrote:
By the way, I know that one can use different packetization times for the same codec, but I’ve never heard, that somebody really uses 30 ms for G711Alaw. Always 20ms.
everything above 60 ms is a nonsense ... and ugly :) It screws your voice quality not even thinking VBD (voice band data) over that line :). Anyhow, Nikolay, your problem is broken IPO h323 stack and the know avaya "flexibility" when interoping with other vendor equipments. Here IPO is unable to negotiate a different framing size than the default and sadly this is the core of the problem. Please, can you send me two tcpdump captures of calls between IPO and FS: 1. a capture with fast start & h245tunneling=true 2. a captire with fast start & h245tunelling=true + h245inSetup I just want to be sure of something. T.
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Tihomir, I've just sent the trace to your gmail address. Nikolay. _____ From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Tihomir Culjaga Sent: Friday, February 12, 2010 11:46 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] h323 - sip call is not working On Fri, Feb 12, 2010 at 3:29 PM, Brian West <brian@freeswitch.org> wrote: This is a rather broad assumption. I have seen 40ms, 60ms and even 80ms in the wild. It all depends on what you want to do. It lowers overhead and increases efficiency on the wire. /b On Feb 12, 2010, at 5:24 AM, Nikolay Kondratyev wrote: By the way, I know that one can use different packetization times for the same codec, but I've never heard, that somebody really uses 30 ms for G711Alaw. Always 20ms. everything above 60 ms is a nonsense ... and ugly :) It screws your voice quality not even thinking VBD (voice band data) over that line :). Anyhow, Nikolay, your problem is broken IPO h323 stack and the know avaya "flexibility" when interoping with other vendor equipments. Here IPO is unable to negotiate a different framing size than the default and sadly this is the core of the problem. Please, can you send me two tcpdump captures of calls between IPO and FS: 1. a capture with fast start & h245tunneling=true 2. a captire with fast start & h245tunelling=true + h245inSetup I just want to be sure of something. T. _______________________________________________ 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
Sent again.. _____ From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Tihomir Culjaga Sent: Monday, February 15, 2010 4:50 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] h323 - sip call is not working On Mon, Feb 15, 2010 at 10:46 AM, Nikolay Kondratyev <kond@nstel.ru> wrote: Tihomir, I've just sent the trace to your gmail address. didn't get anything... Nikolay.
It looks like audio level issue .. can you lower the gain on IPO ? T. On Mon, Feb 15, 2010 at 4:26 PM, Nikolay Kondratyev <kond@nstel.ru> wrote:
Sent again..
------------------------------
*From:* freeswitch-users-bounces@lists.freeswitch.org [mailto: freeswitch-users-bounces@lists.freeswitch.org] *On Behalf Of *Tihomir Culjaga *Sent:* Monday, February 15, 2010 4:50 PM
*To:* freeswitch-users@lists.freeswitch.org *Subject:* Re: [Freeswitch-users] h323 - sip call is not working
On Mon, Feb 15, 2010 at 10:46 AM, Nikolay Kondratyev <kond@nstel.ru> wrote:
Tihomir,
I’ve just sent the trace to your gmail address…
didn't get anything...
Nikolay.
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Mmm. I doubt that it is an audio level problem. Look at the picture that wireshark "voip calls -> player" shows for that wheezy rbt. Here it is: This graph shows that FS really plays "stutter" tone instead of normal rbt. Or am I mistaken? And wireshark analysis of that rtp stream is strange.it shows 22 _seconds_ for that rtp stream.. Here is the text summary of wireshark rtp stream analysis of the stream from FS to IPO: Max delta = 50,65 ms at packet no. 1147 Max jitter = 22556,56 ms. Mean jitter = 238,60 ms. Max skew = 355392,29 ms. Total RTP packets = 1538 (expected 1538) Lost RTP packets = 0 (0,00%) Sequence errors = 0 Duration 36,14 s (513134 ms clock drift, corresponding to 121603 Hz (+1420,03%) I think the problem is on FS site. Thanks and regards, Nikolay. _____ From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Tihomir Culjaga Sent: Monday, February 15, 2010 6:40 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] h323 - sip call is not working It looks like audio level issue .. can you lower the gain on IPO ? T. On Mon, Feb 15, 2010 at 4:26 PM, Nikolay Kondratyev <kond@nstel.ru> wrote: Sent again.. _____ From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Tihomir Culjaga Sent: Monday, February 15, 2010 4:50 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] h323 - sip call is not working On Mon, Feb 15, 2010 at 10:46 AM, Nikolay Kondratyev <kond@nstel.ru> wrote: Tihomir, I've just sent the trace to your gmail address. didn't get anything... Nikolay. _______________________________________________ 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
ok, than try this: edit h323plus/src/h323caps.cxx, grep it for "H323AudioCapability(240, 30) // 240ms max, 30ms desired" ... it should be at line 2598.... replace 30 with 20, recompile (make && make install) make sure you use the new compiled library and start FS. let me know if you still have audio issues.
Tihomir, Thanks a lot, I recompiled h323plus libs as you told me and rbt and voice is ok now. Thanks again, Nikolay. _____ From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Tihomir Culjaga Sent: Monday, February 15, 2010 11:02 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] h323 - sip call is not working ok, than try this: edit h323plus/src/h323caps.cxx, grep it for "H323AudioCapability(240, 30) // 240ms max, 30ms desired" ... it should be at line 2598.... replace 30 with 20, recompile (make && make install) make sure you use the new compiled library and start FS. let me know if you still have audio issues.
Tihomir, I've just sent the tcpdump to your address (because I think mail list will not accept 300K attachement). Thanks and regards, Nikolay. _____ From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Tihomir Culjaga Sent: Wednesday, February 10, 2010 10:14 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] h323 - sip call is not working On Wed, Feb 10, 2010 at 11:49 AM, Nikolay Kondratyev <kond@nstel.ru> wrote: Hi all, I compiled FreeSWITCH Version 1.0.5-20100209-0400 (16587M) with mod_h323. When I call from h323 (Avaya IPOffice) to local fs extention (x-lite) I hear ring back, but when x-lite picks up, he hears silence, while IPOffice user continues to hear ringback. The log is at the http://pastebin.freeswitch.org/12091 My configuration is almost default, several local extentions added, and h323.conf from http://wiki.freeswitch.org/wiki/Mod_h323 5840 - user at IPOffice 2853 - x-lite registered at FS IPOffice ip address: 172.23.14.2 FS ip address 172.23.22.49 Can anybody please advise how to solve that? Is it a configuration or a software problem? Please can you send me the tcpdump as well (not filtered). IPOffice i known to have a "broken" H323 stack. did you try to play with tunneling and h245 in setup settings as well ? It looks like there is some h245 negotiation still pending but cant see that from the logs. Thanks in advance, Nikolay. _______________________________________________ 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
participants (5)
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Brian West
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Georgiewskiy Yuriy
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Nikolay Kondratyev
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Peter Olsson
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Tihomir Culjaga