Each account has it´s own different SIP UDP-Port. Is that possible?
Hello,
My computer is connected to internet via a combined DSL-Router and telephone system (so called FritzBox 7270). On this Fritzbox, there are also some internal Telephones connected (Fritzbox has internaly also a SIP SW-Stack for this telefones).
I would like to write a PJSUA Phone Application on my computer, and to have two accounts - one to the Fritzbox, for internal calls, one to the SIP IP-Provider (for external calls).
Now I get a conflict with Port 5060 (which is also used by Fritzbox), and which can not be forwarded by internal Router.
Using Port 5080 instead (in PJSUA), works fine, as long as I only connect to SIP IP-Provider OR Fritzbox. When using 5080 as port for Account to Fritzbox, there is a conflict (when receiving a call), that the ACK Telegram (which switches SIP phone-call from CONNECTING to CONFIRMED) gets lost in the fritzbox, and after timeout PJSUA disconnects the call ...
The only thing - I could imagine, to make this run - is, to have the SIP-port separated for every account (5060 for Fritzbox, 5080 for external)
What I don´t know: In which way to get separated Port
x A hack in Transport layer, opening two ports, sending telegrams on
the correct port (depending on IP-Adress from the receiver), recei-
ving on both ports ???
x A copy uf UDP-Transport, so having two UDP-Transports inside (seems
to be much work; no easy solution possible).
x Where to read, in which way transport is integrated (and what´s the
difference UDP-Transport and SIP-UDP-Transport)``
x How to figure out, which transport mechanism´s my SIP-Provider is `
able to support?
Who has a tip, in which way it makes sense to proceed ...
With Regards,
Michael
--
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Michael , you must create a different transport (one for each count) and
assign it to that count. Ex:
Creating a transport:
pj_status_t pjStatus;
pjsua_transport_config pjConfig;
char c_strPublicIPAddress[255];
strcpy (c_strPublicIPAddress, strPublicIp.c_str());
pjsua_transport_config_default (&pjConfig);
pjConfig.port = lSipPort; //here you specify the new port (ie:
5060,5061,5062, etc)
pjConfig.public_addr = pj_str(c_strPublicIPAddress);
pjStatus = pjsua_transport_create (PJSIP_TRANSPORT_UDP, &pjConfig,
&m_pjTransportId);
if (pjStatus != PJ_SUCCESS)
{
m_oLogger.write ("CDialerSIP - createtransport() -
pjsua_transport_create - Falló.", CDialerLogger::iLogError);
return false;
}
m_oLogger.write ("CDialerSIP - createtransport() - Ok.",
CDialerLogger::iLogInformation);
m_isTransportCreated = true;
return true;
}
//-----------
Then you must assign that transport to your account. In the following
function, I create an account and assign its transport id to the tansport
that I've created before.
bool CDialerSIP::registeraccount ()
{
pj_status_t pjStatus;
pjsua_acc_config pjConfig;
pj_str_t pjStrId, pjStrRegURL;
pjsua_acc_id pjAccountId;
int iEvent;
pjStrId.ptr = (char*) pj_pool_alloc (m_pjPool, 255);
if (!pjStrId.ptr)
{
m_oLogger.write ("registeraccount - pj_pool_alloc - Falló <pjStrId.ptr>.",
CDialerLogger::iLogError);
return false;
}
pjsua_acc_config_default (&pjConfig);
pj_strcpy (&pjStrId, &(pj_str("sip:")));
pj_strcat (&pjStrId, &m_pjStrSipUser);
pj_strcat (&pjStrId, &(pj_str("@")));
pj_strcat (&pjStrId, &m_pjStrSipDomain);
pjConfig.id = pjStrId;
pjStrRegURL.ptr = (char*) pj_pool_alloc (m_pjPool, 255);
if (!pjStrRegURL.ptr)
{
m_oLogger.write ("registeraccount - pj_pool_alloc - Falló
<pjStrRegURL.ptr>.", CDialerLogger::iLogError);
return false;
}
pj_strcpy2 (&pjStrRegURL, "sip:");
pj_strcat (&pjStrRegURL, &m_pjStrSipDomain);
pjConfig.reg_uri = pjStrRegURL;
pjConfig.cred_count = 1;
pjConfig.cred_info[0].realm = pj_str("*");
pjConfig.cred_info[0].scheme = pj_str("digest");
pjConfig.cred_info[0].username = m_pjStrSipUser;
pjConfig.cred_info[0].data_type = PJSIP_CRED_DATA_PLAIN_PASSWD;
pjConfig.cred_info[0].data = m_pjStrSipPwd;
pjConfig.transport_id = m_pjTransportId; //HERE I ASSIGN THE TRANSPORT ID
THAT I'VE CREATED BEFORE USING pjsua_transport_create.
pjStatus = pjsua_acc_add (&pjConfig, PJ_TRUE, &pjAccountId);
if (pjStatus != PJ_SUCCESS)
{
m_oLogger.write ("registeraccount - pjsua_acc_add - Falló.",
CDialerLogger::iLogError);
return false;
}
pthread_mutex_lock (&m_iMutex_AccountId);
m_pjAccountId = pjAccountId;
pthread_mutex_unlock (&m_iMutex_AccountId);
this->waitevent (&iEvent, false);
if (iEvent != CDialerDialer::i_UserAgent_Register)
{
m_oLogger.write ("registeraccount - Failed.",
CDialerLogger::iLogInformation);
return false;
}
pthread_mutex_lock (&m_iMutex_IsRegistered);
m_isRegistered = true; //la cuenta fue registrada en el sip proxy con
exito. Luego solo se mantendra su estado
// desde la clase CDialerSIPController
pthread_mutex_unlock (&m_iMutex_IsRegistered);
setregisterstate (true);
m_oLogger.write ("registeraccount - Ok.", CDialerLogger::iLogInformation);
return true;
}
that's all.
Let me know if you have a doubt.
bye.
Jose
----- Original Message -----
From: MichaelUndJutta@gmx.net
To: pjsip@lists.pjsip.org
Sent: Thursday, October 07, 2010 9:41 AM
Subject: [SPAM] - [pjsip] Each account has it´s own different SIP UDP-Port.
Is that possible? - Character set not allowed (n/a)
Hello,
My computer is connected to internet via a combined DSL-Router and
telephone system (so called FritzBox 7270). On this Fritzbox, there are
also some internal Telephones connected (Fritzbox has internaly also a SIP
SW-Stack for this telefones).
I would like to write a PJSUA Phone Application on my computer, and to
have two accounts - one to the Fritzbox, for internal calls, one to the
SIP IP-Provider (for external calls).
Now I get a conflict with Port 5060 (which is also used by Fritzbox), and
which can not be forwarded by internal Router.
Using Port 5080 instead (in PJSUA), works fine, as long as I only connect
to SIP IP-Provider OR Fritzbox. When using 5080 as port for Account to
Fritzbox, there is a conflict (when receiving a call), that the ACK
Telegram (which switches SIP phone-call from CONNECTING to CONFIRMED) gets
lost in the fritzbox, and after timeout PJSUA disconnects the call ...
The only thing - I could imagine, to make this run - is, to have the
SIP-port separated for every account (5060 for Fritzbox, 5080 for
external)
What I don´t know: In which way to get separated Port
x A hack in Transport layer, opening two ports, sending telegrams on
the correct port (depending on IP-Adress from the receiver), recei-
ving on both ports ???
x A copy uf UDP-Transport, so having two UDP-Transports inside (seems
to be much work; no easy solution possible).
x Where to read, in which way transport is integrated (and what´s the
difference UDP-Transport and SIP-UDP-Transport)``
x How to figure out, which transport mechanism´s my SIP-Provider is `
able to support?
Who has a tip, in which way it makes sense to proceed ...
With Regards,
Michael
--
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Jetzt freischalten unter http://portal.gmx.net/de/go/maxdome
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