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From: Joegen E. B. <joe...@gm...> - 2008-05-20 23:30:34
|
Whit, Have you tried B2BUpperReg mode instead of full mode? The logs indicate that your invite is relayed (proxy) instead of using the B2BUA. Relay only call does not spawn the RTP Proxy. Joegen Whit Thiele wrote: > > > > > Hey list members, > > > > I have a couple of questions on a simple configuration with OpenSBC that I'm > having trouble proxying the media stream. I've read some other posts on this > topic, but nothing has help so far. Here is the setup: > > > > Office SIP Phone <192.168.0.100> > > | > > [192.168.0.0] Internal Office #1 network > > ---------------------- > > | Office #1 LAN Router | > > ---------------------- > > [25.x.x.x] External IP address of office #1 > > | > > | > > INTERNET > > | > > | > > > > [44.x.x.x] External IP of Office #2 > > ---------------------- > > | Office #2 LAN Router | > > ---------------------- > > [192.168.1.0] Internal Office#2 network > > | > > | > > > > OpenSBC <192.168.1.100> > > | > > | > > Asterisk<192.168.1.101> > > > > > > > > The phone in Office #1 is registering to the Asterisk box in Office #2 using > the UpperRegistration of OpenSBC which is in 'Full Mode'. > > > > > > I am able to register through OpenSBC with the Asterisk box from Office #1, > however I am not getting any media to get proxied via OpenSBC. I've selected > the 'Proxy-All-Media' but nothing seems to work. > > > > I must be missing something simple in the configuration. Where should I > focus my attention? In the B2BUA routes? Proxy-Relay-Routes? > > > > While debugging the SIP messages on the Asterisk output I get the following > SIP message while launching a call: > > > > > > //--------------------------- > > // from Asterisk CLI > > > > SIP/2.0 200 OK > > Via: SIP/2.0/UDP > 192.168.1.100:5060;iid=7375;branch=z9hG4bK74bda256ea24dd1184caf507b23fcaa4-9 > 4d4d1526a6e1a341bd360c9d23def44;uas-addr=192.168.1.101;received=192.168.1.10 > 0;rport=5060 > > Via: SIP/2.0/UDP > 192.168.0.100:61798;branch=z9hG4bK-d87543-77271a19797af374-1--d87543-;rport= > 61798;received=25.X.X.X > > Record-Route: <sip:192.168.1.100:5060;route-shift=192.168.1.100:5060;lr> > > From: "7101" <sip:71...@in...>;tag=114af83a > > To: "5555" <sip:55...@in...>;tag=as262aed0b > > Call-ID: ZTMxZTgzMzFiNDQ0Njk1ODFjNWYxNGJlZDhlMmFhMzY. > > CSeq: 2 INVITE > > User-Agent: Asterisk PBX > > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > > Supported: replaces > > Contact: <sip:5555@192.168.1.101> > > Content-Type: application/sdp > > Content-Length: 268 > > > > v=0 > > o=root 22965 22965 IN IP4 192.168.1.101 > > s=session > > c=IN IP4 192.168.1.101 > > t=0 0 > > m=audio 14662 RTP/AVP 0 8 101 > > a=rtpmap:0 PCMU/8000 > > a=rtpmap:8 PCMA/8000 > > a=rtpmap:101 telephone-event/8000 > > a=fmtp:101 0-16 > > a=silenceSupp:off - - - - > > a=ptime:20 > > a=sendrecv > > //------------------------------- > > > > internalsip.com is an internal mapped DNS to the asterisk box: > > [sip:internalsip.com] sip:192.168.1.101:5060 > > > > > > Debugging Asterisk RTP it shows that its trying to send RTP to 192.168.0.100 > which is the NAT'd address of the office phone in Office #1 which doesn't > make sense to me since obviously it can't reach that network. > > > > > > //----------------- > > // from Asterisk CLI > > > > Sent RTP packet to 192.168.0.100:45976 (type 00, seq 016083, ts 056160, > len 000160) > > Sent RTP packet to 192.168.0.100:45976 (type 00, seq 016084, ts 056320, > len 000160) > > Sent RTP packet to 192.168.0.100:45976 (type 00, seq 016085, ts 056480, > len 000160) > > //------------------ > > > > > > I feel that I'm 98% there... Does anyone have an idea on where to focus my > efforts? > > > > I've attached a b2bualog snip as well in case this helps. > > > > Best Regards, > > > > Whit > > > ------------------------------------------------------------------------ > > ------------------------------------------------------------------------- > This SF.net email is sponsored by: Microsoft > Defy all challenges. Microsoft(R) Visual Studio 2008. > http://clk.atdmt.com/MRT/go/vse0120000070mrt/direct/01/ > ------------------------------------------------------------------------ > > _______________________________________________ > opensipstack-devel mailing list > ope...@li... > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel > ------------------------------------------------------------------------ > > No virus found in this incoming message. > Checked by AVG. > Version: 7.5.524 / Virus Database: 269.23.21/1454 - Release Date: 5/19/2008 7:44 AM |
|
From: Whit T. <de...@wh...> - 2008-05-20 16:45:14
|
Hey list members,
I have a couple of questions on a simple configuration with OpenSBC that I'm
having trouble proxying the media stream. I've read some other posts on this
topic, but nothing has help so far. Here is the setup:
Office SIP Phone <192.168.0.100>
|
[192.168.0.0] Internal Office #1 network
----------------------
| Office #1 LAN Router |
----------------------
[25.x.x.x] External IP address of office #1
|
|
INTERNET
|
|
[44.x.x.x] External IP of Office #2
----------------------
| Office #2 LAN Router |
----------------------
[192.168.1.0] Internal Office#2 network
|
|
OpenSBC <192.168.1.100>
|
|
Asterisk<192.168.1.101>
The phone in Office #1 is registering to the Asterisk box in Office #2 using
the UpperRegistration of OpenSBC which is in 'Full Mode'.
I am able to register through OpenSBC with the Asterisk box from Office #1,
however I am not getting any media to get proxied via OpenSBC. I've selected
the 'Proxy-All-Media' but nothing seems to work.
I must be missing something simple in the configuration. Where should I
focus my attention? In the B2BUA routes? Proxy-Relay-Routes?
While debugging the SIP messages on the Asterisk output I get the following
SIP message while launching a call:
//---------------------------
// from Asterisk CLI
SIP/2.0 200 OK
Via: SIP/2.0/UDP
192.168.1.100:5060;iid=7375;branch=z9hG4bK74bda256ea24dd1184caf507b23fcaa4-9
4d4d1526a6e1a341bd360c9d23def44;uas-addr=192.168.1.101;received=192.168.1.10
0;rport=5060
Via: SIP/2.0/UDP
192.168.0.100:61798;branch=z9hG4bK-d87543-77271a19797af374-1--d87543-;rport=
61798;received=25.X.X.X
Record-Route: <sip:192.168.1.100:5060;route-shift=192.168.1.100:5060;lr>
From: "7101" <sip:71...@in...>;tag=114af83a
To: "5555" <sip:55...@in...>;tag=as262aed0b
Call-ID: ZTMxZTgzMzFiNDQ0Njk1ODFjNWYxNGJlZDhlMmFhMzY.
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:5555@192.168.1.101>
Content-Type: application/sdp
Content-Length: 268
v=0
o=root 22965 22965 IN IP4 192.168.1.101
s=session
c=IN IP4 192.168.1.101
t=0 0
m=audio 14662 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
//-------------------------------
internalsip.com is an internal mapped DNS to the asterisk box:
[sip:internalsip.com] sip:192.168.1.101:5060
Debugging Asterisk RTP it shows that its trying to send RTP to 192.168.0.100
which is the NAT'd address of the office phone in Office #1 which doesn't
make sense to me since obviously it can't reach that network.
//-----------------
// from Asterisk CLI
Sent RTP packet to 192.168.0.100:45976 (type 00, seq 016083, ts 056160,
len 000160)
Sent RTP packet to 192.168.0.100:45976 (type 00, seq 016084, ts 056320,
len 000160)
Sent RTP packet to 192.168.0.100:45976 (type 00, seq 016085, ts 056480,
len 000160)
//------------------
I feel that I'm 98% there... Does anyone have an idea on where to focus my
efforts?
I've attached a b2bualog snip as well in case this helps.
Best Regards,
Whit
|
|
From: prasad k. <pra...@gm...> - 2008-05-20 10:54:58
|
Hi all,ope...@li...
i am using eXosip as sip stack , i am able to initilaze the osip and exosip
and the problem with eXosip _send_register ,where iam not able to fix up the
bug can any bodyhelp me
#include<eXosip2/eXosip.h>
#include<sys/socket.h>
#include<netinet/in.h>
#include<arpa/inet.h>
main() {
int i;
int id;
osip_message_t* reg = NULL;
TRACE_INITIALIZE(6,stdout);
i = eXosip_init();
if(i !=0)
{
printf("Not Initilazed\n");
return -1;
}
else
{
printf("\nInitilazed.....\n");
}
i = eXosip_listen_addr(IPPROTO_UDP,"10.232.19.204
",10234,AF_INET,0);
if(i != 0)
{
eXosip_quit();
printf("Could not initilaize transport layer\n");
return -1;
}
else
{
printf("\nTransport layer initilazed
......................\n");
printf("%d\n", i);
}
eXosip_lock();
id = eXosip_register_build_initial_register("sip:1094@10.232.19.204<sip%3A1094@10.232.19.204>
","10.232.19.204:5060",NULL,1800,®);
printf("%d\n", id);
if(id < 0)
{
eXosip_unlock();
printf("\n unable to build register request\n");
return -1;
}
else
{
printf("Register request built sucessfully\n");
}
i = eXosip_register_build_register(id,1800,reg);
osip_message_set_supported(reg,"100rel");
osip_message_set_supported(reg,"path");
i=eXosip_register_send_register(id,reg);
return -1;
}
Error message i am getting is
Initilazed.....
Transport layer initilazed ......................
0
1
Register request built sucessfully
*** glibc detected *** ./init: double free or corruption (out): 0x097f4cf8
***
======= Backtrace: =========
/lib/libc.so.6[0x166efd]
/lib/libc.so.6(cfree+0x90)[0x16a550]
/usr/lib/libosipparser2.so.3(osip_message_free+0x5c)[0x401f6d]
/usr/lib/libeXosip2.so.4(eXosip_register_send_register+0x112)[0x3acd97]
./init[0x8048772]
/lib/libc.so.6(__libc_start_main+0xdc)[0x116f2c]
./init[0x8048581]
======= Memory map: ========
00101000-00238000 r-xp 00000000 03:01 391121 /lib/libc-2.5.so
00238000-0023a000 r-xp 00137000 03:01 391121 /lib/libc-2.5.so
0023a000-0023b000 rwxp 00139000 03:01 391121 /lib/libc-2.5.so
0023b000-0023e000 rwxp 0023b000 00:00 0
0023e000-00269000 r-xp 00000000 03:01 516459
/usr/lib/libosipparser2.so.2.2.0
00269000-0026a000 rwxp 0002b000 03:01 516459
/usr/lib/libosipparser2.so.2.2.0
002f3000-002fa000 r-xp 00000000 03:01 391124 /lib/librt-2.5.so
002fa000-002fb000 r-xp 00006000 03:01 391124 /lib/librt-2.5.so
002fb000-002fc000 rwxp 00007000 03:01 391124 /lib/librt-2.5
With regards
Prasad
9900782382
|
|
From: Matthias D. <mdr...@gm...> - 2008-05-20 10:17:07
|
Hi everyone, I have played a little bit with sending instant messages with different sip providers. When I send an instant message with my iptel.org account a proxy authentication is required. The problem is that open sip stack seems not to handle this. Regards, M. Dreißig |
|
From: Matthias D. <mdr...@gm...> - 2008-05-19 12:11:14
|
Thanks for your help.
jo...@op... schrieb:
> This is created in the RegisterSessionManager constructor. The registry
> folder serves no purpose for the OSSPhone user agent so you may safely
> remove it in your implementation.
>
> RegistrationDatabase::RegistrationDatabase()
> {
> #if HAS_CPPSQLITE
> m_HasContactRecovery = PrepareContactRecoveryDB(
> "ContactRecovery.sqlite" );
> #else
> m_HasContactRecovery = TRUE;
> OString dir = PProcess::Current().GetFile().GetDirectory() + "registry";
> if( !PFile::Exists( dir.c_str() ) )
> PDirectory::Create( dir.c_str() );
> m_RegRecoveryDIR = dir.c_str();
> #endif
> }
>
>
> Matthias Dreißig wrote:
>
>> Hi all,
>>
>> if I run my application a folder with the name "registry" is created at
>> the location of my application. The folder is empty.
>>
>> Is it possible to prevent the creation of this folder?
>>
>> Regards
>> M. Dreißig
>>
>> -------------------------------------------------------------------------
>> This SF.net email is sponsored by: Microsoft
>> Defy all challenges. Microsoft(R) Visual Studio 2008.
>> http://clk.atdmt.com/MRT/go/vse0120000070mrt/direct/01/
>> _______________________________________________
>> opensipstack-devel mailing list
>> ope...@li...
>> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel
>>
>>
>>
>>
>>
>
>
>
> -------------------------------------------------------------------------
> This SF.net email is sponsored by: Microsoft
> Defy all challenges. Microsoft(R) Visual Studio 2008.
> http://clk.atdmt.com/MRT/go/vse0120000070mrt/direct/01/
> _______________________________________________
> opensipstack-devel mailing list
> ope...@li...
> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel
>
>
|
|
From: Joegen E. B. <joe...@gm...> - 2008-05-17 03:57:05
|
Christian,
I am not sure what you meant by "The only one problem is the syncing of
the data...".
I have implemented a similar class in the past to be used UA that needs
to send file. See the implementation of the VoiceFileChannel class. It
supports RAW input from preencoded G.723.1 and G.729 File. You can use
this class as your reference and create a RAW G.711 handler. You may
then attach it to the OpalMediaStream.
HTH,
Joegen
Christian Wallukat wrote:
> Hi Ilian,
>
>
> I already modified it, so i only use the SIPEndPoint which is derived
> >From OpalOSSEndPoint.
>
> I don’t wanna play files, I get a block of i.e. 2048 bytes of g.711
> Alaw data which I then need to send over the connection ...
>
> If the SIPEndPoint got the answer the RTP will be made and for each
> Direction I get a Stream. If the Stream is opened the callback
>
> BOOL SoftPhoneManager::OnOpenMediaStream(OpalConnection & connection,
> OpalMediaStream & stream)
>
> will be called.
>
> I store the streams inside another class and wait until both streams are
> Active. If they are active I send the data (which comes from a file for
> Test purpose) on the stream where IsSource() == false.
>
> I don’t know what to derive, since all functionality I need is already
> There. The only one problem is the syncing of the data...
>
>
> Attached you find the code I use for testing:
>
>
> void someclass::StreamHandler(void)
> {
> //variables
> ///////////
> OpalMediaStream *pSourceStream = NULL;
> OpalMediaStream *pDestStream = NULL;
> BYTE *pData = NULL;
> int nRead = 0;
> HANDLE hFile =
> CreateFile(_T("c:\\temp\\rec.cal\0"), FILE_WRITE_DATA, FILE_SHARE_READ,
> NULL, CREATE_ALWAYS, FILE_ATTRIBUTE_NORMAL, NULL);
> HANDLE hFile2 =
> CreateFile(_T("c:\\temp\\play.cal\0"), FILE_READ_DATA, FILE_SHARE_READ,
> NULL, OPEN_EXISTING, FILE_ATTRIBUTE_NORMAL, NULL);
> DWORD dwWrite = 0;
> int nReadSize = 0;
> bool bHasBuff = false;
> int nBuff = 0;
> PAdaptiveDelay paDelay;
> CArray carray;
> int nArrayPos = 0;
> PTimeInterval delay;
> RTP_DataFrame packet;
> RTP_DataFrame inpacket;
> RTP_DataFrame pcktout;
> OpalMediaFormat src("G.711-ALaw-64k");
> OpalTranscoder *pTranscoderIn = NULL;
> OpalTranscoder *pTranscoderOut = NULL;
>
>
>
>
> //init play buffers
> for(int n = 0; n < 10; ++n)
> {
> pData = new BYTE[this->m_nDataSize + 10];
> carray.SetAt(n, pData);
> };
>
> while(this->m_bAct == true)
> {
> if(this->m_nStreams > 0)
> {
> if((pDestStream == NULL) || (pSourceStream == NULL))
> {
> //get the source stream
> for(int n = 0; n < this->m_nStreams; ++n)
> {
>
> if(((OpalMediaStream*)this->m_pStreams->GetAt(n))->IsSource() == TRUE)
> {
> pSourceStream =
> (OpalMediaStream*)this->m_pStreams->GetAt(n);
>
> if(pTranscoderIn == NULL)
> {
> //create transcoder
> for convert input
> pTranscoderIn =
> OpalTranscoder::Create(pSourceStream->GetMediaFormat(), src);
> };
> }
> else
> {
> pDestStream =
> (OpalMediaStream*)this->m_pStreams->GetAt(n);
>
> if(pTranscoderOut == NULL)
> {
> //create transcoder
> for convert output
> pTranscoderOut =
> OpalTranscoder::Create(src, pDestStream->GetMediaFormat());
> };
> };
> };
> };
>
> //if the source stream is established
> if(pSourceStream != NULL)
> {
> if(pSourceStream->ReadPacket(inpacket) ==
> TRUE)
> {
> /*
> for(int n = 0; n < nRead; ++n)
> {
> pData[n] =
> bitdreher[pData[n]];
> };
>
> WriteFile(hFile, pData, nRead,
> &dwWrite, NULL);
> */
>
> if(pDestStream != NULL)
> {
> nReadSize =
> pSourceStream->GetDataSize();
> pData =
> (BYTE*)carray.GetAt(nArrayPos++);
>
> if(nArrayPos >= 9)
> {
> nArrayPos = 0;
> };
>
> if((ReadFile(hFile2, pData,
> nReadSize, &dwWrite, NULL) == TRUE) && (dwWrite > 0))
> {
> for(int n = 0; n <
> dwWrite; ++n)
> {
> pData[n] =
> bitdreher[pData[n]];
> };
>
>
> packet.SetPayloadSize(dwWrite);
>
> packet.SetTimestamp(CalculateTimestamp(dwWrite, src));
>
> packet.SetPayloadType(RTP_DataFrame::PCMA);
>
> memcpy(packet.GetPayloadPtr(), pData, dwWrite);
>
> if(pTranscoderOut !=
> NULL)
> {
>
> pTranscoderOut->Convert(packet, pcktout);
>
>
> pDestStream->WritePacket(pcktout);//(pData, dwWrite, nRead);
>
> }
> else
> {
>
>
> ((PDelayChannel*)((OpalRawMediaStream*)pDestStream)->GetChannel())->Write(pD
> ata, dwWrite);
> };
> };
> };
> };
> };
> };
>
> Sleep(10);
> };
>
> CloseHandle(hFile);
> CloseHandle(hFile2);
>
> for(int n = 0; n < 10; ++n)
> {
> pData = (BYTE*)carray.GetAt(n);
> delete pData;
> };
> };
>
>
>
>
> -----Ursprüngliche Nachricht-----
> Von: ope...@li...
> [mailto:ope...@li...] Im Auftrag von
> Ilian Jeri C. Pinzon
> Gesendet: Donnerstag, 15. Mai 2008 12:21
> An: ope...@li...
> Betreff: Re: [OpenSIPStack] Playing Data on a mediastream / synchronize data
>
> Hi,
>
> You mentioned earlier that you wanted to play to or record from a
> source/target other than the PC's sound system? The Softphone is using
> the PC sound system endpoint (i.e. OpalPCSSEndPoint) by default. But
> since you mentioned that your target environment may not necessarily
> have a mic or speaker, you will need to change that.
>
> If for example you need to play to or record from a file, you will need
> to create classes like OpalFileEndPoint, OpalFileConnection, and
> OpalFileMediaStream which are analogous to OpalPCSSEndPoint,
> OpalPCSSConnection, and OpalAudioMediaStream. Just to refer to the said
> classes and pattern your implementation to their interfaces.
>
> Then use your OpalFileEndPoint instance as as endpoint in SoftPhone.
>
> Christian Wallukat wrote:
>
>> Hi Ilian,
>>
>>
>> Thanks for the response, but I don't understand where to override it ?
>>
>> I use the SoftPhone code to get it running and this is my last problem.
>> The MediaStream was already created if this function
>>
>> BOOL SoftPhoneManager::OnOpenMediaStream(OpalConnection & connection,
>> OpalMediaStream & stream)
>>
>> Is called.
>>
>> Do you mean, that I need to change it in OpalConnection or where should I
>>
> do
>
>> so ?
>>
>>
>> Kind regards
>>
>>
>> Christian
>>
>> -----Ursprüngliche Nachricht-----
>> Von: ope...@li...
>> [mailto:ope...@li...] Im Auftrag von
>> Ilian Jeri C. Pinzon
>> Gesendet: Donnerstag, 15. Mai 2008 05:51
>> An: ope...@li...
>> Betreff: Re: [OpenSIPStack] Playing Data on a mediastream / synchronize
>>
> data
>
>> Hi,
>>
>> Christian Wallukat wrote:
>>
>>
>>> Hi all,
>>>
>>>
>>> I try to send G711 Alaw data on the destination media stream but it
>>> Is snatchy if the delay between the packets doesn’t match...
>>>
>>> The delay of 20ms is not always right, how can I time the packets ?
>>>
>>> I use OpalMediaStream::WriteFrame() for sending the data ...
>>>
>>>
>>>
>> The OpalMediaPatch thread should do this for you automatically. You
>> don't have to call WriteFrame(). What you just need to do is implement
>> your derived OpalMediaStream interfaces properly (if they are not yet
>> implemented that is).
>>
>>
>>> Kind regards
>>>
>>>
>>> Christian
>>>
>>> No virus found in this outgoing message.
>>> Checked by AVG.
>>> Version: 7.5.524 / Virus Database: 269.23.11/1422 - Release Date:
>>>
>>>
>> 08.05.2008
>>
>>
>>> 17:24
>>>
>>>
>>>
>>> -------------------------------------------------------------------------
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>>>
>>>
>> 5/13/2008 7:55 PM
>>
>>
>>>
>>>
>>>
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>>
> 08.05.2008
>
>> 17:24
>>
>>
>> No virus found in this outgoing message.
>> Checked by AVG.
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> 08.05.2008
>
>> 17:24
>>
>>
>>
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>>
>>
>>
>>
>
>
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> 17:24
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> 07:24
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|
|
From: <jo...@op...> - 2008-05-17 03:40:40
|
This is created in the RegisterSessionManager constructor. The registry
folder serves no purpose for the OSSPhone user agent so you may safely
remove it in your implementation.
RegistrationDatabase::RegistrationDatabase()
{
#if HAS_CPPSQLITE
m_HasContactRecovery = PrepareContactRecoveryDB(
"ContactRecovery.sqlite" );
#else
m_HasContactRecovery = TRUE;
OString dir = PProcess::Current().GetFile().GetDirectory() + "registry";
if( !PFile::Exists( dir.c_str() ) )
PDirectory::Create( dir.c_str() );
m_RegRecoveryDIR = dir.c_str();
#endif
}
Matthias Dreißig wrote:
> Hi all,
>
> if I run my application a folder with the name "registry" is created at
> the location of my application. The folder is empty.
>
> Is it possible to prevent the creation of this folder?
>
> Regards
> M. Dreißig
>
> -------------------------------------------------------------------------
> This SF.net email is sponsored by: Microsoft
> Defy all challenges. Microsoft(R) Visual Studio 2008.
> http://clk.atdmt.com/MRT/go/vse0120000070mrt/direct/01/
> _______________________________________________
> opensipstack-devel mailing list
> ope...@li...
> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel
>
>
>
>
|
|
From: Christian W. <cwa...@gm...> - 2008-05-16 07:43:34
|
Hi Ilian,
I already modified it, so i only use the SIPEndPoint which is derived
>From OpalOSSEndPoint.
I dont wanna play files, I get a block of i.e. 2048 bytes of g.711
Alaw data which I then need to send over the connection ...
If the SIPEndPoint got the answer the RTP will be made and for each
Direction I get a Stream. If the Stream is opened the callback
BOOL SoftPhoneManager::OnOpenMediaStream(OpalConnection & connection,
OpalMediaStream & stream)
will be called.
I store the streams inside another class and wait until both streams are
Active. If they are active I send the data (which comes from a file for
Test purpose) on the stream where IsSource() == false.
I dont know what to derive, since all functionality I need is already
There. The only one problem is the syncing of the data...
Attached you find the code I use for testing:
void someclass::StreamHandler(void)
{
//variables
///////////
OpalMediaStream *pSourceStream = NULL;
OpalMediaStream *pDestStream = NULL;
BYTE *pData = NULL;
int nRead = 0;
HANDLE hFile =
CreateFile(_T("c:\\temp\\rec.cal\0"), FILE_WRITE_DATA, FILE_SHARE_READ,
NULL, CREATE_ALWAYS, FILE_ATTRIBUTE_NORMAL, NULL);
HANDLE hFile2 =
CreateFile(_T("c:\\temp\\play.cal\0"), FILE_READ_DATA, FILE_SHARE_READ,
NULL, OPEN_EXISTING, FILE_ATTRIBUTE_NORMAL, NULL);
DWORD dwWrite = 0;
int nReadSize = 0;
bool bHasBuff = false;
int nBuff = 0;
PAdaptiveDelay paDelay;
CArray carray;
int nArrayPos = 0;
PTimeInterval delay;
RTP_DataFrame packet;
RTP_DataFrame inpacket;
RTP_DataFrame pcktout;
OpalMediaFormat src("G.711-ALaw-64k");
OpalTranscoder *pTranscoderIn = NULL;
OpalTranscoder *pTranscoderOut = NULL;
//init play buffers
for(int n = 0; n < 10; ++n)
{
pData = new BYTE[this->m_nDataSize + 10];
carray.SetAt(n, pData);
};
while(this->m_bAct == true)
{
if(this->m_nStreams > 0)
{
if((pDestStream == NULL) || (pSourceStream == NULL))
{
//get the source stream
for(int n = 0; n < this->m_nStreams; ++n)
{
if(((OpalMediaStream*)this->m_pStreams->GetAt(n))->IsSource() == TRUE)
{
pSourceStream =
(OpalMediaStream*)this->m_pStreams->GetAt(n);
if(pTranscoderIn == NULL)
{
//create transcoder
for convert input
pTranscoderIn =
OpalTranscoder::Create(pSourceStream->GetMediaFormat(), src);
};
}
else
{
pDestStream =
(OpalMediaStream*)this->m_pStreams->GetAt(n);
if(pTranscoderOut == NULL)
{
//create transcoder
for convert output
pTranscoderOut =
OpalTranscoder::Create(src, pDestStream->GetMediaFormat());
};
};
};
};
//if the source stream is established
if(pSourceStream != NULL)
{
if(pSourceStream->ReadPacket(inpacket) ==
TRUE)
{
/*
for(int n = 0; n < nRead; ++n)
{
pData[n] =
bitdreher[pData[n]];
};
WriteFile(hFile, pData, nRead,
&dwWrite, NULL);
*/
if(pDestStream != NULL)
{
nReadSize =
pSourceStream->GetDataSize();
pData =
(BYTE*)carray.GetAt(nArrayPos++);
if(nArrayPos >= 9)
{
nArrayPos = 0;
};
if((ReadFile(hFile2, pData,
nReadSize, &dwWrite, NULL) == TRUE) && (dwWrite > 0))
{
for(int n = 0; n <
dwWrite; ++n)
{
pData[n] =
bitdreher[pData[n]];
};
packet.SetPayloadSize(dwWrite);
packet.SetTimestamp(CalculateTimestamp(dwWrite, src));
packet.SetPayloadType(RTP_DataFrame::PCMA);
memcpy(packet.GetPayloadPtr(), pData, dwWrite);
if(pTranscoderOut !=
NULL)
{
pTranscoderOut->Convert(packet, pcktout);
pDestStream->WritePacket(pcktout);//(pData, dwWrite, nRead);
}
else
{
((PDelayChannel*)((OpalRawMediaStream*)pDestStream)->GetChannel())->Write(pD
ata, dwWrite);
};
};
};
};
};
};
Sleep(10);
};
CloseHandle(hFile);
CloseHandle(hFile2);
for(int n = 0; n < 10; ++n)
{
pData = (BYTE*)carray.GetAt(n);
delete pData;
};
};
-----Ursprüngliche Nachricht-----
Von: ope...@li...
[mailto:ope...@li...] Im Auftrag von
Ilian Jeri C. Pinzon
Gesendet: Donnerstag, 15. Mai 2008 12:21
An: ope...@li...
Betreff: Re: [OpenSIPStack] Playing Data on a mediastream / synchronize data
Hi,
You mentioned earlier that you wanted to play to or record from a
source/target other than the PC's sound system? The Softphone is using
the PC sound system endpoint (i.e. OpalPCSSEndPoint) by default. But
since you mentioned that your target environment may not necessarily
have a mic or speaker, you will need to change that.
If for example you need to play to or record from a file, you will need
to create classes like OpalFileEndPoint, OpalFileConnection, and
OpalFileMediaStream which are analogous to OpalPCSSEndPoint,
OpalPCSSConnection, and OpalAudioMediaStream. Just to refer to the said
classes and pattern your implementation to their interfaces.
Then use your OpalFileEndPoint instance as as endpoint in SoftPhone.
Christian Wallukat wrote:
> Hi Ilian,
>
>
> Thanks for the response, but I don't understand where to override it ?
>
> I use the SoftPhone code to get it running and this is my last problem.
> The MediaStream was already created if this function
>
> BOOL SoftPhoneManager::OnOpenMediaStream(OpalConnection & connection,
> OpalMediaStream & stream)
>
> Is called.
>
> Do you mean, that I need to change it in OpalConnection or where should I
do
> so ?
>
>
> Kind regards
>
>
> Christian
>
> -----Ursprüngliche Nachricht-----
> Von: ope...@li...
> [mailto:ope...@li...] Im Auftrag von
> Ilian Jeri C. Pinzon
> Gesendet: Donnerstag, 15. Mai 2008 05:51
> An: ope...@li...
> Betreff: Re: [OpenSIPStack] Playing Data on a mediastream / synchronize
data
>
> Hi,
>
> Christian Wallukat wrote:
>
>> Hi all,
>>
>>
>> I try to send G711 Alaw data on the destination media stream but it
>> Is snatchy if the delay between the packets doesnt match...
>>
>> The delay of 20ms is not always right, how can I time the packets ?
>>
>> I use OpalMediaStream::WriteFrame() for sending the data ...
>>
>>
> The OpalMediaPatch thread should do this for you automatically. You
> don't have to call WriteFrame(). What you just need to do is implement
> your derived OpalMediaStream interfaces properly (if they are not yet
> implemented that is).
>
>> Kind regards
>>
>>
>> Christian
>>
>> No virus found in this outgoing message.
>> Checked by AVG.
>> Version: 7.5.524 / Virus Database: 269.23.11/1422 - Release Date:
>>
> 08.05.2008
>
>> 17:24
>>
>>
>>
>> -------------------------------------------------------------------------
>> This SF.net email is sponsored by: Microsoft
>> Defy all challenges. Microsoft(R) Visual Studio 2008.
>> http://clk.atdmt.com/MRT/go/vse0120000070mrt/direct/01/
>> _______________________________________________
>> opensipstack-devel mailing list
>> ope...@li...
>> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel
>>
>> ------------------------------------------------------------------------
>>
>> No virus found in this incoming message.
>> Checked by AVG.
>> Version: 7.5.524 / Virus Database: 269.23.16/1431 - Release Date:
>>
> 5/13/2008 7:55 PM
>
>>
>>
>
>
> -------------------------------------------------------------------------
> This SF.net email is sponsored by: Microsoft
> Defy all challenges. Microsoft(R) Visual Studio 2008.
> http://clk.atdmt.com/MRT/go/vse0120000070mrt/direct/01/
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> ope...@li...
> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel
>
> No virus found in this incoming message.
> Checked by AVG.
> Version: 7.5.524 / Virus Database: 269.23.11/1422 - Release Date:
08.05.2008
> 17:24
>
>
> No virus found in this outgoing message.
> Checked by AVG.
> Version: 7.5.524 / Virus Database: 269.23.11/1422 - Release Date:
08.05.2008
> 17:24
>
>
>
> -------------------------------------------------------------------------
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>
-------------------------------------------------------------------------
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_______________________________________________
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ope...@li...
https://lists.sourceforge.net/lists/listinfo/opensipstack-devel
No virus found in this incoming message.
Checked by AVG.
Version: 7.5.524 / Virus Database: 269.23.11/1422 - Release Date: 08.05.2008
17:24
No virus found in this outgoing message.
Checked by AVG.
Version: 7.5.524 / Virus Database: 269.23.16/1434 - Release Date: 15.05.2008
07:24
|
|
From: Matthias D. <mdr...@gm...> - 2008-05-16 05:32:43
|
By the way to use my own RegisterSessionManager and CallSessionManager I had to change the members of OpalOSSUserAgent from private to protected. All other classes have protected members. Regards M. Dreißig Joegen E. Baclor schrieb: > Matthias Dreißig wrote: > >> Thanks for fast your answer. >> >> And if I want to adapt the registration I have to implement a subclass >> of RegisterSessionManager and RegisterSession, am I right? >> >> > > Right. If you need to modify the behavior of the RegisterSession, then > you may also subclass it. However, if you don't need to, just > sub-classing RegisterSessionManager would be ok too. All the necessary > callbacks for authentication of each session is already exposed in the > manager. Take a look at OpalOSSEndPoint.cxx specifically > OpalOSSRegistrar class for a sample implementation. > > >> Regards, >> M. Dreißig >> >> jo...@op... schrieb: >> >> >>> You will need to implement your own subclass of CallSessionManager and >>> CallSession for you to be able to do this cleanly. >>> >>> You need to implement your own override of >>> >>> BOOL CallSession::MakeCall( >>> const SIPURI & uri, >>> const OString & sdp >>> ); >>> >>> >>> It might be possible to do it in OPAL but you will need to modify the >>> behavior of the media callbacks to ignore none existence of media >>> channels. I wouldn't advise it. >>> >>> >>> Matthias Dreißig wrote: >>> >>> >>> >>>> Hi everyone, >>>> >>>> is it possible to set a content type when making a call with the help of >>>> opal manager? I want to set my own content type and body. >>>> >>>> Regards, >>>> M. Dreißig >>>> >>>> ------------------------------------------------------------------------- >>>> This SF.net email is sponsored by the 2008 JavaOne(SM) Conference >>>> Don't miss this year's exciting event. There's still time to save $100. >>>> Use priority code J8TL2D2. >>>> http://ad.doubleclick.net/clk;198757673;13503038;p?http://java.sun.com/javaone >>>> _______________________________________________ >>>> opensipstack-devel mailing list >>>> ope...@li... >>>> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>> ------------------------------------------------------------------------- >>> This SF.net email is sponsored by the 2008 JavaOne(SM) Conference >>> Don't miss this year's exciting event. There's still time to save $100. >>> Use priority code J8TL2D2. >>> http://ad.doubleclick.net/clk;198757673;13503038;p?http://java.sun.com/javaone >>> _______________________________________________ >>> opensipstack-devel mailing list >>> ope...@li... >>> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel >>> >>> >>> >>> >> ------------------------------------------------------------------------- >> This SF.net email is sponsored by the 2008 JavaOne(SM) Conference >> Don't miss this year's exciting event. There's still time to save $100. >> Use priority code J8TL2D2. >> http://ad.doubleclick.net/clk;198757673;13503038;p?http://java.sun.com/javaone >> _______________________________________________ >> opensipstack-devel mailing list >> ope...@li... >> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel >> >> >> >> > > > > ------------------------------------------------------------------------- > This SF.net email is sponsored by the 2008 JavaOne(SM) Conference > Don't miss this year's exciting event. There's still time to save $100. > Use priority code J8TL2D2. > http://ad.doubleclick.net/clk;198757673;13503038;p?http://java.sun.com/javaone > _______________________________________________ > opensipstack-devel mailing list > ope...@li... > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel > > |
|
From: Matthias D. <mdr...@gm...> - 2008-05-15 19:41:31
|
Hi all, if I run my application a folder with the name "registry" is created at the location of my application. The folder is empty. Is it possible to prevent the creation of this folder? Regards M. Dreißig |
|
From: Matthias D. <mdr...@gm...> - 2008-05-15 19:25:51
|
Hi,
I successful did my changes, thanks for your help.
Now I have the problem that on shutting down the application the library
is not terminated correctly (clean). There are five threads that are
terminated with exit code 1. That are the housekeeper and the garbage
collector threads. On shut down I call already
ClearAllCalls();
if( sipEndPoint != NULL ) {
sipEndPoint->GetUserAgent()->GetStack().Terminate();
}
// Shut down the cleaner thread
garbageCollectExit.Signal();
garbageCollector->WaitForTermination();
Can someone help me?
Thanks.
M.Dreißig
Joegen E. Baclor schrieb:
> Matthias Dreißig wrote:
>
>> Thanks for fast your answer.
>>
>> And if I want to adapt the registration I have to implement a subclass
>> of RegisterSessionManager and RegisterSession, am I right?
>>
>>
>
> Right. If you need to modify the behavior of the RegisterSession, then
> you may also subclass it. However, if you don't need to, just
> sub-classing RegisterSessionManager would be ok too. All the necessary
> callbacks for authentication of each session is already exposed in the
> manager. Take a look at OpalOSSEndPoint.cxx specifically
> OpalOSSRegistrar class for a sample implementation.
>
>
>> Regards,
>> M. Dreißig
>>
>> jo...@op... schrieb:
>>
>>
>>> You will need to implement your own subclass of CallSessionManager and
>>> CallSession for you to be able to do this cleanly.
>>>
>>> You need to implement your own override of
>>>
>>> BOOL CallSession::MakeCall(
>>> const SIPURI & uri,
>>> const OString & sdp
>>> );
>>>
>>>
>>> It might be possible to do it in OPAL but you will need to modify the
>>> behavior of the media callbacks to ignore none existence of media
>>> channels. I wouldn't advise it.
>>>
>>>
>>> Matthias Dreißig wrote:
>>>
>>>
>>>
>>>> Hi everyone,
>>>>
>>>> is it possible to set a content type when making a call with the help of
>>>> opal manager? I want to set my own content type and body.
>>>>
>>>> Regards,
>>>> M. Dreißig
>>>>
>>>> -------------------------------------------------------------------------
>>>> This SF.net email is sponsored by the 2008 JavaOne(SM) Conference
>>>> Don't miss this year's exciting event. There's still time to save $100.
>>>> Use priority code J8TL2D2.
>>>> http://ad.doubleclick.net/clk;198757673;13503038;p?http://java.sun.com/javaone
>>>> _______________________________________________
>>>> opensipstack-devel mailing list
>>>> ope...@li...
>>>> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>> -------------------------------------------------------------------------
>>> This SF.net email is sponsored by the 2008 JavaOne(SM) Conference
>>> Don't miss this year's exciting event. There's still time to save $100.
>>> Use priority code J8TL2D2.
>>> http://ad.doubleclick.net/clk;198757673;13503038;p?http://java.sun.com/javaone
>>> _______________________________________________
>>> opensipstack-devel mailing list
>>> ope...@li...
>>> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel
>>>
>>>
>>>
>>>
>> -------------------------------------------------------------------------
>> This SF.net email is sponsored by the 2008 JavaOne(SM) Conference
>> Don't miss this year's exciting event. There's still time to save $100.
>> Use priority code J8TL2D2.
>> http://ad.doubleclick.net/clk;198757673;13503038;p?http://java.sun.com/javaone
>> _______________________________________________
>> opensipstack-devel mailing list
>> ope...@li...
>> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel
>>
>>
>>
>>
>
>
>
> -------------------------------------------------------------------------
> This SF.net email is sponsored by the 2008 JavaOne(SM) Conference
> Don't miss this year's exciting event. There's still time to save $100.
> Use priority code J8TL2D2.
> http://ad.doubleclick.net/clk;198757673;13503038;p?http://java.sun.com/javaone
> _______________________________________________
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> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel
>
>
|
|
From: Ilian J. C. P. <ip...@so...> - 2008-05-15 10:21:19
|
Hi, You mentioned earlier that you wanted to play to or record from a source/target other than the PC's sound system? The Softphone is using the PC sound system endpoint (i.e. OpalPCSSEndPoint) by default. But since you mentioned that your target environment may not necessarily have a mic or speaker, you will need to change that. If for example you need to play to or record from a file, you will need to create classes like OpalFileEndPoint, OpalFileConnection, and OpalFileMediaStream which are analogous to OpalPCSSEndPoint, OpalPCSSConnection, and OpalAudioMediaStream. Just to refer to the said classes and pattern your implementation to their interfaces. Then use your OpalFileEndPoint instance as as endpoint in SoftPhone. Christian Wallukat wrote: > Hi Ilian, > > > Thanks for the response, but I don't understand where to override it ? > > I use the SoftPhone code to get it running and this is my last problem. > The MediaStream was already created if this function > > BOOL SoftPhoneManager::OnOpenMediaStream(OpalConnection & connection, > OpalMediaStream & stream) > > Is called. > > Do you mean, that I need to change it in OpalConnection or where should I do > so ? > > > Kind regards > > > Christian > > -----Ursprüngliche Nachricht----- > Von: ope...@li... > [mailto:ope...@li...] Im Auftrag von > Ilian Jeri C. Pinzon > Gesendet: Donnerstag, 15. Mai 2008 05:51 > An: ope...@li... > Betreff: Re: [OpenSIPStack] Playing Data on a mediastream / synchronize data > > Hi, > > Christian Wallukat wrote: > >> Hi all, >> >> >> I try to send G711 Alaw data on the destination media stream but it >> Is snatchy if the delay between the packets doesn’t match... >> >> The delay of 20ms is not always right, how can I time the packets ? >> >> I use OpalMediaStream::WriteFrame() for sending the data ... >> >> > The OpalMediaPatch thread should do this for you automatically. You > don't have to call WriteFrame(). What you just need to do is implement > your derived OpalMediaStream interfaces properly (if they are not yet > implemented that is). > >> Kind regards >> >> >> Christian >> >> No virus found in this outgoing message. >> Checked by AVG. >> Version: 7.5.524 / Virus Database: 269.23.11/1422 - Release Date: >> > 08.05.2008 > >> 17:24 >> >> >> >> ------------------------------------------------------------------------- >> This SF.net email is sponsored by: Microsoft >> Defy all challenges. Microsoft(R) Visual Studio 2008. >> http://clk.atdmt.com/MRT/go/vse0120000070mrt/direct/01/ >> _______________________________________________ >> opensipstack-devel mailing list >> ope...@li... >> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel >> >> ------------------------------------------------------------------------ >> >> No virus found in this incoming message. >> Checked by AVG. >> Version: 7.5.524 / Virus Database: 269.23.16/1431 - Release Date: >> > 5/13/2008 7:55 PM > >> >> > > > ------------------------------------------------------------------------- > This SF.net email is sponsored by: Microsoft > Defy all challenges. Microsoft(R) Visual Studio 2008. > http://clk.atdmt.com/MRT/go/vse0120000070mrt/direct/01/ > _______________________________________________ > opensipstack-devel mailing list > ope...@li... > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel > > No virus found in this incoming message. > Checked by AVG. > Version: 7.5.524 / Virus Database: 269.23.11/1422 - Release Date: 08.05.2008 > 17:24 > > > No virus found in this outgoing message. > Checked by AVG. > Version: 7.5.524 / Virus Database: 269.23.11/1422 - Release Date: 08.05.2008 > 17:24 > > > > ------------------------------------------------------------------------- > This SF.net email is sponsored by: Microsoft > Defy all challenges. Microsoft(R) Visual Studio 2008. > http://clk.atdmt.com/MRT/go/vse0120000070mrt/direct/01/ > _______________________________________________ > opensipstack-devel mailing list > ope...@li... > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel > > > |
|
From: Christian W. <cwa...@gm...> - 2008-05-15 09:49:52
|
Hi Ilian, Thanks for the response, but I don't understand where to override it ? I use the SoftPhone code to get it running and this is my last problem. The MediaStream was already created if this function BOOL SoftPhoneManager::OnOpenMediaStream(OpalConnection & connection, OpalMediaStream & stream) Is called. Do you mean, that I need to change it in OpalConnection or where should I do so ? Kind regards Christian -----Ursprüngliche Nachricht----- Von: ope...@li... [mailto:ope...@li...] Im Auftrag von Ilian Jeri C. Pinzon Gesendet: Donnerstag, 15. Mai 2008 05:51 An: ope...@li... Betreff: Re: [OpenSIPStack] Playing Data on a mediastream / synchronize data Hi, Christian Wallukat wrote: > Hi all, > > > I try to send G711 Alaw data on the destination media stream but it > Is snatchy if the delay between the packets doesnt match... > > The delay of 20ms is not always right, how can I time the packets ? > > I use OpalMediaStream::WriteFrame() for sending the data ... > The OpalMediaPatch thread should do this for you automatically. You don't have to call WriteFrame(). What you just need to do is implement your derived OpalMediaStream interfaces properly (if they are not yet implemented that is). > > > Kind regards > > > Christian > > No virus found in this outgoing message. > Checked by AVG. > Version: 7.5.524 / Virus Database: 269.23.11/1422 - Release Date: 08.05.2008 > 17:24 > > > > ------------------------------------------------------------------------- > This SF.net email is sponsored by: Microsoft > Defy all challenges. Microsoft(R) Visual Studio 2008. > http://clk.atdmt.com/MRT/go/vse0120000070mrt/direct/01/ > _______________________________________________ > opensipstack-devel mailing list > ope...@li... > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel > > ------------------------------------------------------------------------ > > No virus found in this incoming message. > Checked by AVG. > Version: 7.5.524 / Virus Database: 269.23.16/1431 - Release Date: 5/13/2008 7:55 PM > ------------------------------------------------------------------------- This SF.net email is sponsored by: Microsoft Defy all challenges. Microsoft(R) Visual Studio 2008. http://clk.atdmt.com/MRT/go/vse0120000070mrt/direct/01/ _______________________________________________ opensipstack-devel mailing list ope...@li... https://lists.sourceforge.net/lists/listinfo/opensipstack-devel No virus found in this incoming message. Checked by AVG. Version: 7.5.524 / Virus Database: 269.23.11/1422 - Release Date: 08.05.2008 17:24 No virus found in this outgoing message. Checked by AVG. Version: 7.5.524 / Virus Database: 269.23.11/1422 - Release Date: 08.05.2008 17:24 |
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From: Ilian J. C. P. <ip...@so...> - 2008-05-15 09:23:46
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I cannot reproduce the issue you reported using the source code version in CVS head. The are many modifications committed to the CVS version. The issue you're reporting may be fixed already. Please test again using the said version. Tamas Csomor wrote: > Hi, > > could you write me, how it will work in my case? > > Regards, > Tamas Csomor > > On Thu, May 15, 2008 at 9:01 AM, Ilian Jeri C. Pinzon < > ip...@so...> wrote: > > >> Hi, >> >> Please use the version from CVS head. CVS instructions are in: >> >> http://opensipstack.org/cvs.html >> >> Tamas Csomor wrote: >> >>> Hy, >>> >>> I'm new here, and i have a little problem. I want to use the OSSPhone in >>> >> our >> >>> university by testing the SIP message via PC->Huawei IMS. The client can >>> >> not >> >>> login to the IMS. I think, when the IMS send back the authorization >>> >> message >> >>> there is a nonce"anything" line, but when the client send back, in this >>> message isn't... >>> Can anybodyhelp me, how can i fix it in the source code? >>> Thanx for the help. >>> Best regards >>> Tamas Csomor >>> >>> ----------------11:37:29.780---------------- >>> SEND: enc=0 534 Bytes to 10.0.1.1:5060:UDP (REGISTER sip:mik.bme.huSIP/2.0) >>> Interface Address= >>> REGISTER sip:mik.bme.hu SIP/2.0 >>> From: test04 <sip:te...@mi... <sip%3At...@mi...> < >>> >> sip%3At...@mi... <sip%253...@mi...>> >> >>>> ;tag=9ce0bacc8efb18109a15901176cd0c70 >>>> >>>> >>> To: sip:te...@mi... <sip%3At...@mi...> < >>> >> sip%3At...@mi... <sip%253...@mi...>> >> >>> Via: SIP/2.0/UDP 152.66.87.157:5061 >>> ;iid=5406;branch=z9hG4bK9ce0bacc8efb18109a16901176cd0c70;uas-addr= >>> >> 10.0.1.1 >> >>> ;rport >>> CSeq: 1 REGISTER >>> Call-ID: 9ce0bacc-8efb-1810-9a5e-901176cd0c70 >>> Contact: "test04" <sip:test04@152.66.87.157:5061;transport=udp> >>> User-Agent: OpenSIPStack-1.1.7-7 >>> Expires: 3600 >>> Max-Forwards: 10 >>> Allow: INVITE, BYE, ACK, REFER, MESSAGE, INFO, NOTIFY, OPTIONS >>> Content-Length: 0 >>> >>> ----------------11:37:29.994---------------- >>> SEND: enc=0 534 Bytes to 10.0.1.1:5060:UDP (REGISTER sip:mik.bme.huSIP/2.0) >>> Interface Address= >>> REGISTER sip:mik.bme.hu SIP/2.0 >>> From: test04 <sip:te...@mi... <sip%3At...@mi...> < >>> >> sip%3At...@mi... <sip%253...@mi...>> >> >>>> ;tag=9ce0bacc8efb18109a15901176cd0c70 >>>> >>>> >>> To: sip:te...@mi... <sip%3At...@mi...> < >>> >> sip%3At...@mi... <sip%253...@mi...>> >> >>> Via: SIP/2.0/UDP 152.66.87.157:5061 >>> ;iid=5406;branch=z9hG4bK9ce0bacc8efb18109a16901176cd0c70;uas-addr= >>> >> 10.0.1.1 >> >>> ;rport >>> CSeq: 1 REGISTER >>> Call-ID: 9ce0bacc-8efb-1810-9a5e-901176cd0c70 >>> Contact: "test04" <sip:test04@152.66.87.157:5061;transport=udp> >>> User-Agent: OpenSIPStack-1.1.7-7 >>> Expires: 3600 >>> Max-Forwards: 10 >>> Allow: INVITE, BYE, ACK, REFER, MESSAGE, INFO, NOTIFY, OPTIONS >>> Content-Length: 0 >>> >>> ----------------11:37:30.090---------------- >>> RCV: enc=0 446 Bytes from RCVADDR: 10.0.1.1:RCVPORT: 5060:UDP (SIP/2.0 >>> >> 401 >> >>> Unauthorized) >>> SIP/2.0 401 Unauthorized >>> From: test04 <sip:te...@mi... <sip%3At...@mi...> < >>> >> sip%3At...@mi... <sip%253...@mi...>> >> >>>> ;tag=9ce0bacc8efb18109a15901176cd0c70 >>>> >>>> >>> To: <sip:te...@mi... <sip%3At...@mi...> < >>> >> sip%3At...@mi... <sip%253...@mi...>>>;tag=37a9a0a6 >> >>> Via: SIP/2.0/UDP 152.66.87.157:5061 >>> ;iid=5406;branch=z9hG4bK9ce0bacc8efb18109a16901176cd0c70;uas-addr= >>> >> 10.0.1.1 >> >>> ;rport=5061 >>> CSeq: 1 REGISTER >>> Call-ID: 9ce0bacc-8efb-1810-9a5e-901176cd0c70 >>> WWW-Authenticate: Digest realm="mik.bme.hu", >>> nonce="Yig2Ky1Zz+1NNWE00IIEWw=", algorithm=MD5 //here is >>> the nonce, that the client not send back >>> Content-Length: 0 >>> >>> ----------------11:37:30.100---------------- >>> SEND: enc=0 698 Bytes to 10.0.1.1:5060:UDP (REGISTER sip:mik.bme.huSIP/2.0) >>> Interface Address= >>> REGISTER sip:mik.bme.hu SIP/2.0 >>> From: test04 <sip:te...@mi... <sip%3At...@mi...> < >>> >> sip%3At...@mi... <sip%253...@mi...>> >> >>>> ;tag=c960bbcc8efb18109a16901176cd0c70 >>>> >>>> >>> To: sip:te...@mi... <sip%3At...@mi...> < >>> >> sip%3At...@mi... <sip%253...@mi...>> >> >>> Via: SIP/2.0/UDP 152.66.87.157:5061 >>> ;iid=5406;branch=z9hG4bKc960bbcc8efb18109a17901176cd0c70;uas-addr= >>> >> 10.0.1.1 >> >>> ;rport >>> CSeq: 2 REGISTER >>> Call-ID: 9ce0bacc-8efb-1810-9a5e-901176cd0c70 >>> Contact: "test04" <sip:test04@152.66.87.157:5061;transport=udp> >>> User-Agent: OpenSIPStack-1.1.7-7 >>> Expires: 3600 >>> Max-Forwards: 10 >>> Authorization: Digest username="te...@mi...", realm="mik.bme.hu", >>> nonce="", uri="sip:mik.bme.hu", >>> >> response="87d50c014f0cfc437d5baf83a4e77478", >> >>> algorithm=MD5 //here nonce is empty >>> Allow: INVITE, BYE, ACK, REFER, MESSAGE, INFO, NOTIFY, OPTIONS >>> Content-Length: 0 >>> >>> ----------------11:37:30.236---------------- >>> RCV: enc=0 427 Bytes from RCVADDR: 10.0.1.1:RCVPORT: 5060:UDP (SIP/2.0 >>> >> 403 >> >>> Forbidden) >>> SIP/2.0 403 Forbidden >>> From: test04 <sip:te...@mi... <sip%3At...@mi...> < >>> >> sip%3At...@mi... <sip%253...@mi...>> >> >>>> ;tag=c960bbcc8efb18109a16901176cd0c70 >>>> >>>> >>> To: <sip:te...@mi... <sip%3At...@mi...> < >>> >> sip%3At...@mi... <sip%253...@mi...>>>;tag=7f107ba9 >> >>> Via: SIP/2.0/UDP 152.66.87.157:5061 >>> ;iid=5406;branch=z9hG4bKc960bbcc8efb18109a17901176cd0c70;uas-addr= >>> >> 10.0.1.1 >> >>> ;rport=5061 >>> CSeq: 2 REGISTER >>> Call-ID: 9ce0bacc-8efb-1810-9a5e-901176cd0c70 >>> Warning: 399 0173402704.S.207.11.24.mik.bme.hu.00106 "Authentication >>> failure" >>> Content-Length: 0 >>> ------------------------------------------------------------------------- >>> This SF.net email is sponsored by: Microsoft >>> Defy all challenges. Microsoft(R) Visual Studio 2008. >>> http://clk.atdmt.com/MRT/go/vse0120000070mrt/direct/01/ >>> _______________________________________________ >>> opensipstack-devel mailing list >>> ope...@li... >>> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel >>> >>> >>> >>> >> ------------------------------------------------------------------------- >> This SF.net email is sponsored by: Microsoft >> Defy all challenges. Microsoft(R) Visual Studio 2008. >> http://clk.atdmt.com/MRT/go/vse0120000070mrt/direct/01/ >> _______________________________________________ >> opensipstack-devel mailing list >> ope...@li... >> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel >> >> > ------------------------------------------------------------------------- > This SF.net email is sponsored by: Microsoft > Defy all challenges. Microsoft(R) Visual Studio 2008. > http://clk.atdmt.com/MRT/go/vse0120000070mrt/direct/01/ > _______________________________________________ > opensipstack-devel mailing list > ope...@li... > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel > > > |
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From: Tamas C. <cso...@gm...> - 2008-05-15 07:16:45
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Hi, could you write me, how it will work in my case? Regards, Tamas Csomor On Thu, May 15, 2008 at 9:01 AM, Ilian Jeri C. Pinzon < ip...@so...> wrote: > Hi, > > Please use the version from CVS head. CVS instructions are in: > > http://opensipstack.org/cvs.html > > Tamas Csomor wrote: > > Hy, > > > > I'm new here, and i have a little problem. I want to use the OSSPhone in > our > > university by testing the SIP message via PC->Huawei IMS. The client can > not > > login to the IMS. I think, when the IMS send back the authorization > message > > there is a nonce"anything" line, but when the client send back, in this > > message isn't... > > Can anybodyhelp me, how can i fix it in the source code? > > Thanx for the help. > > Best regards > > Tamas Csomor > > > > ----------------11:37:29.780---------------- > > SEND: enc=0 534 Bytes to 10.0.1.1:5060:UDP (REGISTER sip:mik.bme.huSIP/2.0) > > Interface Address= > > REGISTER sip:mik.bme.hu SIP/2.0 > > From: test04 <sip:te...@mi... <sip%3At...@mi...> < > sip%3At...@mi... <sip%253...@mi...>> > > > >> ;tag=9ce0bacc8efb18109a15901176cd0c70 > >> > > To: sip:te...@mi... <sip%3At...@mi...> < > sip%3At...@mi... <sip%253...@mi...>> > > Via: SIP/2.0/UDP 152.66.87.157:5061 > > ;iid=5406;branch=z9hG4bK9ce0bacc8efb18109a16901176cd0c70;uas-addr= > 10.0.1.1 > > ;rport > > CSeq: 1 REGISTER > > Call-ID: 9ce0bacc-8efb-1810-9a5e-901176cd0c70 > > Contact: "test04" <sip:test04@152.66.87.157:5061;transport=udp> > > User-Agent: OpenSIPStack-1.1.7-7 > > Expires: 3600 > > Max-Forwards: 10 > > Allow: INVITE, BYE, ACK, REFER, MESSAGE, INFO, NOTIFY, OPTIONS > > Content-Length: 0 > > > > ----------------11:37:29.994---------------- > > SEND: enc=0 534 Bytes to 10.0.1.1:5060:UDP (REGISTER sip:mik.bme.huSIP/2.0) > > Interface Address= > > REGISTER sip:mik.bme.hu SIP/2.0 > > From: test04 <sip:te...@mi... <sip%3At...@mi...> < > sip%3At...@mi... <sip%253...@mi...>> > > > >> ;tag=9ce0bacc8efb18109a15901176cd0c70 > >> > > To: sip:te...@mi... <sip%3At...@mi...> < > sip%3At...@mi... <sip%253...@mi...>> > > Via: SIP/2.0/UDP 152.66.87.157:5061 > > ;iid=5406;branch=z9hG4bK9ce0bacc8efb18109a16901176cd0c70;uas-addr= > 10.0.1.1 > > ;rport > > CSeq: 1 REGISTER > > Call-ID: 9ce0bacc-8efb-1810-9a5e-901176cd0c70 > > Contact: "test04" <sip:test04@152.66.87.157:5061;transport=udp> > > User-Agent: OpenSIPStack-1.1.7-7 > > Expires: 3600 > > Max-Forwards: 10 > > Allow: INVITE, BYE, ACK, REFER, MESSAGE, INFO, NOTIFY, OPTIONS > > Content-Length: 0 > > > > ----------------11:37:30.090---------------- > > RCV: enc=0 446 Bytes from RCVADDR: 10.0.1.1:RCVPORT: 5060:UDP (SIP/2.0 > 401 > > Unauthorized) > > SIP/2.0 401 Unauthorized > > From: test04 <sip:te...@mi... <sip%3At...@mi...> < > sip%3At...@mi... <sip%253...@mi...>> > > > >> ;tag=9ce0bacc8efb18109a15901176cd0c70 > >> > > To: <sip:te...@mi... <sip%3At...@mi...> < > sip%3At...@mi... <sip%253...@mi...>>>;tag=37a9a0a6 > > Via: SIP/2.0/UDP 152.66.87.157:5061 > > ;iid=5406;branch=z9hG4bK9ce0bacc8efb18109a16901176cd0c70;uas-addr= > 10.0.1.1 > > ;rport=5061 > > CSeq: 1 REGISTER > > Call-ID: 9ce0bacc-8efb-1810-9a5e-901176cd0c70 > > WWW-Authenticate: Digest realm="mik.bme.hu", > > nonce="Yig2Ky1Zz+1NNWE00IIEWw=", algorithm=MD5 //here is > > the nonce, that the client not send back > > Content-Length: 0 > > > > ----------------11:37:30.100---------------- > > SEND: enc=0 698 Bytes to 10.0.1.1:5060:UDP (REGISTER sip:mik.bme.huSIP/2.0) > > Interface Address= > > REGISTER sip:mik.bme.hu SIP/2.0 > > From: test04 <sip:te...@mi... <sip%3At...@mi...> < > sip%3At...@mi... <sip%253...@mi...>> > > > >> ;tag=c960bbcc8efb18109a16901176cd0c70 > >> > > To: sip:te...@mi... <sip%3At...@mi...> < > sip%3At...@mi... <sip%253...@mi...>> > > Via: SIP/2.0/UDP 152.66.87.157:5061 > > ;iid=5406;branch=z9hG4bKc960bbcc8efb18109a17901176cd0c70;uas-addr= > 10.0.1.1 > > ;rport > > CSeq: 2 REGISTER > > Call-ID: 9ce0bacc-8efb-1810-9a5e-901176cd0c70 > > Contact: "test04" <sip:test04@152.66.87.157:5061;transport=udp> > > User-Agent: OpenSIPStack-1.1.7-7 > > Expires: 3600 > > Max-Forwards: 10 > > Authorization: Digest username="te...@mi...", realm="mik.bme.hu", > > nonce="", uri="sip:mik.bme.hu", > response="87d50c014f0cfc437d5baf83a4e77478", > > algorithm=MD5 //here nonce is empty > > Allow: INVITE, BYE, ACK, REFER, MESSAGE, INFO, NOTIFY, OPTIONS > > Content-Length: 0 > > > > ----------------11:37:30.236---------------- > > RCV: enc=0 427 Bytes from RCVADDR: 10.0.1.1:RCVPORT: 5060:UDP (SIP/2.0 > 403 > > Forbidden) > > SIP/2.0 403 Forbidden > > From: test04 <sip:te...@mi... <sip%3At...@mi...> < > sip%3At...@mi... <sip%253...@mi...>> > > > >> ;tag=c960bbcc8efb18109a16901176cd0c70 > >> > > To: <sip:te...@mi... <sip%3At...@mi...> < > sip%3At...@mi... <sip%253...@mi...>>>;tag=7f107ba9 > > Via: SIP/2.0/UDP 152.66.87.157:5061 > > ;iid=5406;branch=z9hG4bKc960bbcc8efb18109a17901176cd0c70;uas-addr= > 10.0.1.1 > > ;rport=5061 > > CSeq: 2 REGISTER > > Call-ID: 9ce0bacc-8efb-1810-9a5e-901176cd0c70 > > Warning: 399 0173402704.S.207.11.24.mik.bme.hu.00106 "Authentication > > failure" > > Content-Length: 0 > > ------------------------------------------------------------------------- > > This SF.net email is sponsored by: Microsoft > > Defy all challenges. Microsoft(R) Visual Studio 2008. > > http://clk.atdmt.com/MRT/go/vse0120000070mrt/direct/01/ > > _______________________________________________ > > opensipstack-devel mailing list > > ope...@li... > > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel > > > > > > > > > ------------------------------------------------------------------------- > This SF.net email is sponsored by: Microsoft > Defy all challenges. Microsoft(R) Visual Studio 2008. > http://clk.atdmt.com/MRT/go/vse0120000070mrt/direct/01/ > _______________________________________________ > opensipstack-devel mailing list > ope...@li... > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel > |
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From: Joegen E. B. <joe...@gm...> - 2008-05-15 07:14:15
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Wow, I think you found a bug in the parser. I will guess it's the "=" character inside the nonce. We will confirm this and advise the list when a patch is ready in CVS. Joegen Tamas Csomor wrote: > Hy, > > I'm new here, and i have a little problem. I want to use the OSSPhone in our > university by testing the SIP message via PC->Huawei IMS. The client can not > login to the IMS. I think, when the IMS send back the authorization message > there is a nonce"anything" line, but when the client send back, in this > message isn't... > Can anybodyhelp me, how can i fix it in the source code? > Thanx for the help. > Best regards > Tamas Csomor > > ----------------11:37:29.780---------------- > SEND: enc=0 534 Bytes to 10.0.1.1:5060:UDP (REGISTER sip:mik.bme.hu SIP/2.0) > Interface Address= > REGISTER sip:mik.bme.hu SIP/2.0 > From: test04 <sip:te...@mi... <sip%3At...@mi...> > >> ;tag=9ce0bacc8efb18109a15901176cd0c70 >> > To: sip:te...@mi... <sip%3At...@mi...> > Via: SIP/2.0/UDP 152.66.87.157:5061 > ;iid=5406;branch=z9hG4bK9ce0bacc8efb18109a16901176cd0c70;uas-addr=10.0.1.1 > ;rport > CSeq: 1 REGISTER > Call-ID: 9ce0bacc-8efb-1810-9a5e-901176cd0c70 > Contact: "test04" <sip:test04@152.66.87.157:5061;transport=udp> > User-Agent: OpenSIPStack-1.1.7-7 > Expires: 3600 > Max-Forwards: 10 > Allow: INVITE, BYE, ACK, REFER, MESSAGE, INFO, NOTIFY, OPTIONS > Content-Length: 0 > > ----------------11:37:29.994---------------- > SEND: enc=0 534 Bytes to 10.0.1.1:5060:UDP (REGISTER sip:mik.bme.hu SIP/2.0) > Interface Address= > REGISTER sip:mik.bme.hu SIP/2.0 > From: test04 <sip:te...@mi... <sip%3At...@mi...> > >> ;tag=9ce0bacc8efb18109a15901176cd0c70 >> > To: sip:te...@mi... <sip%3At...@mi...> > Via: SIP/2.0/UDP 152.66.87.157:5061 > ;iid=5406;branch=z9hG4bK9ce0bacc8efb18109a16901176cd0c70;uas-addr=10.0.1.1 > ;rport > CSeq: 1 REGISTER > Call-ID: 9ce0bacc-8efb-1810-9a5e-901176cd0c70 > Contact: "test04" <sip:test04@152.66.87.157:5061;transport=udp> > User-Agent: OpenSIPStack-1.1.7-7 > Expires: 3600 > Max-Forwards: 10 > Allow: INVITE, BYE, ACK, REFER, MESSAGE, INFO, NOTIFY, OPTIONS > Content-Length: 0 > > ----------------11:37:30.090---------------- > RCV: enc=0 446 Bytes from RCVADDR: 10.0.1.1:RCVPORT: 5060:UDP (SIP/2.0 401 > Unauthorized) > SIP/2.0 401 Unauthorized > From: test04 <sip:te...@mi... <sip%3At...@mi...> > >> ;tag=9ce0bacc8efb18109a15901176cd0c70 >> > To: <sip:te...@mi... <sip%3At...@mi...>>;tag=37a9a0a6 > Via: SIP/2.0/UDP 152.66.87.157:5061 > ;iid=5406;branch=z9hG4bK9ce0bacc8efb18109a16901176cd0c70;uas-addr=10.0.1.1 > ;rport=5061 > CSeq: 1 REGISTER > Call-ID: 9ce0bacc-8efb-1810-9a5e-901176cd0c70 > WWW-Authenticate: Digest realm="mik.bme.hu", > nonce="Yig2Ky1Zz+1NNWE00IIEWw=", algorithm=MD5 //here is > the nonce, that the client not send back > Content-Length: 0 > > ----------------11:37:30.100---------------- > SEND: enc=0 698 Bytes to 10.0.1.1:5060:UDP (REGISTER sip:mik.bme.hu SIP/2.0) > Interface Address= > REGISTER sip:mik.bme.hu SIP/2.0 > From: test04 <sip:te...@mi... <sip%3At...@mi...> > >> ;tag=c960bbcc8efb18109a16901176cd0c70 >> > To: sip:te...@mi... <sip%3At...@mi...> > Via: SIP/2.0/UDP 152.66.87.157:5061 > ;iid=5406;branch=z9hG4bKc960bbcc8efb18109a17901176cd0c70;uas-addr=10.0.1.1 > ;rport > CSeq: 2 REGISTER > Call-ID: 9ce0bacc-8efb-1810-9a5e-901176cd0c70 > Contact: "test04" <sip:test04@152.66.87.157:5061;transport=udp> > User-Agent: OpenSIPStack-1.1.7-7 > Expires: 3600 > Max-Forwards: 10 > Authorization: Digest username="te...@mi...", realm="mik.bme.hu", > nonce="", uri="sip:mik.bme.hu", response="87d50c014f0cfc437d5baf83a4e77478", > algorithm=MD5 //here nonce is empty > Allow: INVITE, BYE, ACK, REFER, MESSAGE, INFO, NOTIFY, OPTIONS > Content-Length: 0 > > ----------------11:37:30.236---------------- > RCV: enc=0 427 Bytes from RCVADDR: 10.0.1.1:RCVPORT: 5060:UDP (SIP/2.0 403 > Forbidden) > SIP/2.0 403 Forbidden > From: test04 <sip:te...@mi... <sip%3At...@mi...> > >> ;tag=c960bbcc8efb18109a16901176cd0c70 >> > To: <sip:te...@mi... <sip%3At...@mi...>>;tag=7f107ba9 > Via: SIP/2.0/UDP 152.66.87.157:5061 > ;iid=5406;branch=z9hG4bKc960bbcc8efb18109a17901176cd0c70;uas-addr=10.0.1.1 > ;rport=5061 > CSeq: 2 REGISTER > Call-ID: 9ce0bacc-8efb-1810-9a5e-901176cd0c70 > Warning: 399 0173402704.S.207.11.24.mik.bme.hu.00106 "Authentication > failure" > Content-Length: 0 > ------------------------------------------------------------------------- > This SF.net email is sponsored by: Microsoft > Defy all challenges. Microsoft(R) Visual Studio 2008. > http://clk.atdmt.com/MRT/go/vse0120000070mrt/direct/01/ > _______________________________________________ > opensipstack-devel mailing list > ope...@li... > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel > > > |
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From: Ilian J. C. P. <ip...@so...> - 2008-05-15 07:01:56
|
Hi, Please use the version from CVS head. CVS instructions are in: http://opensipstack.org/cvs.html Tamas Csomor wrote: > Hy, > > I'm new here, and i have a little problem. I want to use the OSSPhone in our > university by testing the SIP message via PC->Huawei IMS. The client can not > login to the IMS. I think, when the IMS send back the authorization message > there is a nonce"anything" line, but when the client send back, in this > message isn't... > Can anybodyhelp me, how can i fix it in the source code? > Thanx for the help. > Best regards > Tamas Csomor > > ----------------11:37:29.780---------------- > SEND: enc=0 534 Bytes to 10.0.1.1:5060:UDP (REGISTER sip:mik.bme.hu SIP/2.0) > Interface Address= > REGISTER sip:mik.bme.hu SIP/2.0 > From: test04 <sip:te...@mi... <sip%3At...@mi...> > >> ;tag=9ce0bacc8efb18109a15901176cd0c70 >> > To: sip:te...@mi... <sip%3At...@mi...> > Via: SIP/2.0/UDP 152.66.87.157:5061 > ;iid=5406;branch=z9hG4bK9ce0bacc8efb18109a16901176cd0c70;uas-addr=10.0.1.1 > ;rport > CSeq: 1 REGISTER > Call-ID: 9ce0bacc-8efb-1810-9a5e-901176cd0c70 > Contact: "test04" <sip:test04@152.66.87.157:5061;transport=udp> > User-Agent: OpenSIPStack-1.1.7-7 > Expires: 3600 > Max-Forwards: 10 > Allow: INVITE, BYE, ACK, REFER, MESSAGE, INFO, NOTIFY, OPTIONS > Content-Length: 0 > > ----------------11:37:29.994---------------- > SEND: enc=0 534 Bytes to 10.0.1.1:5060:UDP (REGISTER sip:mik.bme.hu SIP/2.0) > Interface Address= > REGISTER sip:mik.bme.hu SIP/2.0 > From: test04 <sip:te...@mi... <sip%3At...@mi...> > >> ;tag=9ce0bacc8efb18109a15901176cd0c70 >> > To: sip:te...@mi... <sip%3At...@mi...> > Via: SIP/2.0/UDP 152.66.87.157:5061 > ;iid=5406;branch=z9hG4bK9ce0bacc8efb18109a16901176cd0c70;uas-addr=10.0.1.1 > ;rport > CSeq: 1 REGISTER > Call-ID: 9ce0bacc-8efb-1810-9a5e-901176cd0c70 > Contact: "test04" <sip:test04@152.66.87.157:5061;transport=udp> > User-Agent: OpenSIPStack-1.1.7-7 > Expires: 3600 > Max-Forwards: 10 > Allow: INVITE, BYE, ACK, REFER, MESSAGE, INFO, NOTIFY, OPTIONS > Content-Length: 0 > > ----------------11:37:30.090---------------- > RCV: enc=0 446 Bytes from RCVADDR: 10.0.1.1:RCVPORT: 5060:UDP (SIP/2.0 401 > Unauthorized) > SIP/2.0 401 Unauthorized > From: test04 <sip:te...@mi... <sip%3At...@mi...> > >> ;tag=9ce0bacc8efb18109a15901176cd0c70 >> > To: <sip:te...@mi... <sip%3At...@mi...>>;tag=37a9a0a6 > Via: SIP/2.0/UDP 152.66.87.157:5061 > ;iid=5406;branch=z9hG4bK9ce0bacc8efb18109a16901176cd0c70;uas-addr=10.0.1.1 > ;rport=5061 > CSeq: 1 REGISTER > Call-ID: 9ce0bacc-8efb-1810-9a5e-901176cd0c70 > WWW-Authenticate: Digest realm="mik.bme.hu", > nonce="Yig2Ky1Zz+1NNWE00IIEWw=", algorithm=MD5 //here is > the nonce, that the client not send back > Content-Length: 0 > > ----------------11:37:30.100---------------- > SEND: enc=0 698 Bytes to 10.0.1.1:5060:UDP (REGISTER sip:mik.bme.hu SIP/2.0) > Interface Address= > REGISTER sip:mik.bme.hu SIP/2.0 > From: test04 <sip:te...@mi... <sip%3At...@mi...> > >> ;tag=c960bbcc8efb18109a16901176cd0c70 >> > To: sip:te...@mi... <sip%3At...@mi...> > Via: SIP/2.0/UDP 152.66.87.157:5061 > ;iid=5406;branch=z9hG4bKc960bbcc8efb18109a17901176cd0c70;uas-addr=10.0.1.1 > ;rport > CSeq: 2 REGISTER > Call-ID: 9ce0bacc-8efb-1810-9a5e-901176cd0c70 > Contact: "test04" <sip:test04@152.66.87.157:5061;transport=udp> > User-Agent: OpenSIPStack-1.1.7-7 > Expires: 3600 > Max-Forwards: 10 > Authorization: Digest username="te...@mi...", realm="mik.bme.hu", > nonce="", uri="sip:mik.bme.hu", response="87d50c014f0cfc437d5baf83a4e77478", > algorithm=MD5 //here nonce is empty > Allow: INVITE, BYE, ACK, REFER, MESSAGE, INFO, NOTIFY, OPTIONS > Content-Length: 0 > > ----------------11:37:30.236---------------- > RCV: enc=0 427 Bytes from RCVADDR: 10.0.1.1:RCVPORT: 5060:UDP (SIP/2.0 403 > Forbidden) > SIP/2.0 403 Forbidden > From: test04 <sip:te...@mi... <sip%3At...@mi...> > >> ;tag=c960bbcc8efb18109a16901176cd0c70 >> > To: <sip:te...@mi... <sip%3At...@mi...>>;tag=7f107ba9 > Via: SIP/2.0/UDP 152.66.87.157:5061 > ;iid=5406;branch=z9hG4bKc960bbcc8efb18109a17901176cd0c70;uas-addr=10.0.1.1 > ;rport=5061 > CSeq: 2 REGISTER > Call-ID: 9ce0bacc-8efb-1810-9a5e-901176cd0c70 > Warning: 399 0173402704.S.207.11.24.mik.bme.hu.00106 "Authentication > failure" > Content-Length: 0 > ------------------------------------------------------------------------- > This SF.net email is sponsored by: Microsoft > Defy all challenges. Microsoft(R) Visual Studio 2008. > http://clk.atdmt.com/MRT/go/vse0120000070mrt/direct/01/ > _______________________________________________ > opensipstack-devel mailing list > ope...@li... > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel > > > |
|
From: Tamas C. <cso...@gm...> - 2008-05-15 06:48:48
|
Hy,
I'm new here, and i have a little problem. I want to use the OSSPhone in our
university by testing the SIP message via PC->Huawei IMS. The client can not
login to the IMS. I think, when the IMS send back the authorization message
there is a nonce"anything" line, but when the client send back, in this
message isn't...
Can anybodyhelp me, how can i fix it in the source code?
Thanx for the help.
Best regards
Tamas Csomor
----------------11:37:29.780----------------
SEND: enc=0 534 Bytes to 10.0.1.1:5060:UDP (REGISTER sip:mik.bme.hu SIP/2.0)
Interface Address=
REGISTER sip:mik.bme.hu SIP/2.0
From: test04 <sip:te...@mi... <sip%3At...@mi...>
>;tag=9ce0bacc8efb18109a15901176cd0c70
To: sip:te...@mi... <sip%3At...@mi...>
Via: SIP/2.0/UDP 152.66.87.157:5061
;iid=5406;branch=z9hG4bK9ce0bacc8efb18109a16901176cd0c70;uas-addr=10.0.1.1
;rport
CSeq: 1 REGISTER
Call-ID: 9ce0bacc-8efb-1810-9a5e-901176cd0c70
Contact: "test04" <sip:test04@152.66.87.157:5061;transport=udp>
User-Agent: OpenSIPStack-1.1.7-7
Expires: 3600
Max-Forwards: 10
Allow: INVITE, BYE, ACK, REFER, MESSAGE, INFO, NOTIFY, OPTIONS
Content-Length: 0
----------------11:37:29.994----------------
SEND: enc=0 534 Bytes to 10.0.1.1:5060:UDP (REGISTER sip:mik.bme.hu SIP/2.0)
Interface Address=
REGISTER sip:mik.bme.hu SIP/2.0
From: test04 <sip:te...@mi... <sip%3At...@mi...>
>;tag=9ce0bacc8efb18109a15901176cd0c70
To: sip:te...@mi... <sip%3At...@mi...>
Via: SIP/2.0/UDP 152.66.87.157:5061
;iid=5406;branch=z9hG4bK9ce0bacc8efb18109a16901176cd0c70;uas-addr=10.0.1.1
;rport
CSeq: 1 REGISTER
Call-ID: 9ce0bacc-8efb-1810-9a5e-901176cd0c70
Contact: "test04" <sip:test04@152.66.87.157:5061;transport=udp>
User-Agent: OpenSIPStack-1.1.7-7
Expires: 3600
Max-Forwards: 10
Allow: INVITE, BYE, ACK, REFER, MESSAGE, INFO, NOTIFY, OPTIONS
Content-Length: 0
----------------11:37:30.090----------------
RCV: enc=0 446 Bytes from RCVADDR: 10.0.1.1:RCVPORT: 5060:UDP (SIP/2.0 401
Unauthorized)
SIP/2.0 401 Unauthorized
From: test04 <sip:te...@mi... <sip%3At...@mi...>
>;tag=9ce0bacc8efb18109a15901176cd0c70
To: <sip:te...@mi... <sip%3At...@mi...>>;tag=37a9a0a6
Via: SIP/2.0/UDP 152.66.87.157:5061
;iid=5406;branch=z9hG4bK9ce0bacc8efb18109a16901176cd0c70;uas-addr=10.0.1.1
;rport=5061
CSeq: 1 REGISTER
Call-ID: 9ce0bacc-8efb-1810-9a5e-901176cd0c70
WWW-Authenticate: Digest realm="mik.bme.hu",
nonce="Yig2Ky1Zz+1NNWE00IIEWw=", algorithm=MD5 //here is
the nonce, that the client not send back
Content-Length: 0
----------------11:37:30.100----------------
SEND: enc=0 698 Bytes to 10.0.1.1:5060:UDP (REGISTER sip:mik.bme.hu SIP/2.0)
Interface Address=
REGISTER sip:mik.bme.hu SIP/2.0
From: test04 <sip:te...@mi... <sip%3At...@mi...>
>;tag=c960bbcc8efb18109a16901176cd0c70
To: sip:te...@mi... <sip%3At...@mi...>
Via: SIP/2.0/UDP 152.66.87.157:5061
;iid=5406;branch=z9hG4bKc960bbcc8efb18109a17901176cd0c70;uas-addr=10.0.1.1
;rport
CSeq: 2 REGISTER
Call-ID: 9ce0bacc-8efb-1810-9a5e-901176cd0c70
Contact: "test04" <sip:test04@152.66.87.157:5061;transport=udp>
User-Agent: OpenSIPStack-1.1.7-7
Expires: 3600
Max-Forwards: 10
Authorization: Digest username="te...@mi...", realm="mik.bme.hu",
nonce="", uri="sip:mik.bme.hu", response="87d50c014f0cfc437d5baf83a4e77478",
algorithm=MD5 //here nonce is empty
Allow: INVITE, BYE, ACK, REFER, MESSAGE, INFO, NOTIFY, OPTIONS
Content-Length: 0
----------------11:37:30.236----------------
RCV: enc=0 427 Bytes from RCVADDR: 10.0.1.1:RCVPORT: 5060:UDP (SIP/2.0 403
Forbidden)
SIP/2.0 403 Forbidden
From: test04 <sip:te...@mi... <sip%3At...@mi...>
>;tag=c960bbcc8efb18109a16901176cd0c70
To: <sip:te...@mi... <sip%3At...@mi...>>;tag=7f107ba9
Via: SIP/2.0/UDP 152.66.87.157:5061
;iid=5406;branch=z9hG4bKc960bbcc8efb18109a17901176cd0c70;uas-addr=10.0.1.1
;rport=5061
CSeq: 2 REGISTER
Call-ID: 9ce0bacc-8efb-1810-9a5e-901176cd0c70
Warning: 399 0173402704.S.207.11.24.mik.bme.hu.00106 "Authentication
failure"
Content-Length: 0
|
|
From: Ilian J. C. P. <ip...@so...> - 2008-05-15 03:51:02
|
Hi, Christian Wallukat wrote: > Hi all, > > > I try to send G711 Alaw data on the destination media stream but it > Is snatchy if the delay between the packets doesn’t match... > > The delay of 20ms is not always right, how can I time the packets ? > > I use OpalMediaStream::WriteFrame() for sending the data ... > The OpalMediaPatch thread should do this for you automatically. You don't have to call WriteFrame(). What you just need to do is implement your derived OpalMediaStream interfaces properly (if they are not yet implemented that is). > > > Kind regards > > > Christian > > No virus found in this outgoing message. > Checked by AVG. > Version: 7.5.524 / Virus Database: 269.23.11/1422 - Release Date: 08.05.2008 > 17:24 > > > > ------------------------------------------------------------------------- > This SF.net email is sponsored by: Microsoft > Defy all challenges. Microsoft(R) Visual Studio 2008. > http://clk.atdmt.com/MRT/go/vse0120000070mrt/direct/01/ > _______________________________________________ > opensipstack-devel mailing list > ope...@li... > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel > > ------------------------------------------------------------------------ > > No virus found in this incoming message. > Checked by AVG. > Version: 7.5.524 / Virus Database: 269.23.16/1431 - Release Date: 5/13/2008 7:55 PM > |
|
From: OpenSource C. <ope...@gm...> - 2008-05-14 20:13:22
|
Hello all, I am new to SBC stuff, please pardon my naive questions. May I know what is the meaning of the osbc modes ? Is it standard for an SBC to support these modes ? ~Abna |
|
From: Christian W. <cwa...@gm...> - 2008-05-14 16:17:54
|
Hi all, I try to send G711 Alaw data on the destination media stream but it Is snatchy if the delay between the packets doesn’t match... The delay of 20ms is not always right, how can I time the packets ? I use OpalMediaStream::WriteFrame() for sending the data ... Kind regards Christian No virus found in this outgoing message. Checked by AVG. Version: 7.5.524 / Virus Database: 269.23.11/1422 - Release Date: 08.05.2008 17:24 |
|
From: H.Kropf <mai...@gl...> - 2008-05-14 13:46:47
|
How to set custom User-Agent in SoftPhone or SoftPhoneInterface? Is it possible to use short mode of SIP-tags (f:, t: instead of From:, To:) in SoftPhone or SoftPhoneInterface? |
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From: H.Kropf <mai...@gl...> - 2008-05-14 13:39:18
|
>Joegen E. Baclor wrote:
>There is no need for you to cal
>SendCancel() directly. Simply calling HangUpCall() will either send a
>BYE for mid-dialog and a CANCEL during setup stages.
Sorry.
"Mea culpa" ::)
I casually used it
SetUpCall("pc:*", dest, SetCurrentCallToken() );
instead of
SetUpCall("pc:*", dest, m_CurrentCallToken);
> Joegen E. Baclor wrote:
> I will patch the stack accordingly.
Probably OpenSER does not send message CANCEL to the client at whom it
is not declared CANCEL in list "Allow" of messages REGISTER and INVITE.
The sessions rejected by the CalledStation therefore hang.
|
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From: Joegen E. B. <joe...@gm...> - 2008-05-14 04:36:30
|
H.Kropf wrote: > I wish to cancel an outgoing call which the remote side does not answer > longtime (ringin on remote side). For this purpose I should send message > CANCEL. > For this it is necessary to call method CallSession:: SendCancel()? > But how to obtain CallSession object? > Which module are you referring to? I am assuming you are referring to the SoftPhoneInterface module? There is no need for you to cal SendCancel() directly. Simply calling HangUpCall() will either send a BYE for mid-dialog and a CANCEL during setup stages. > Some softphones (for example EyeBeam set) announce method CANCEL in the > list "Allow" of message INVITE. Why OSS does not do it? > I have re-read RFC3261 and ACK and CANCEL are mandatory in the Allow header. I was quick to assume that INVITE in allow header implies support for ACK and CANCEL. I will patch the stack accordingly. Thanks for pointing it out. > ------------------------------------------------------------------------- > This SF.net email is sponsored by: Microsoft > Defy all challenges. Microsoft(R) Visual Studio 2008. > http://clk.atdmt.com/MRT/go/vse0120000070mrt/direct/01/ > _______________________________________________ > opensipstack-devel mailing list > ope...@li... > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel > > > |
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From: H.Kropf <mai...@gl...> - 2008-05-13 14:55:15
|
I wish to cancel an outgoing call which the remote side does not answer longtime (ringin on remote side). For this purpose I should send message CANCEL. For this it is necessary to call method CallSession:: SendCancel()? But how to obtain CallSession object? Some softphones (for example EyeBeam set) announce method CANCEL in the list "Allow" of message INVITE. Why OSS does not do it? |
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