(... re-posting, the original post doesn't seem to have made it into the list - my apologies, in case of a double-posting)
Hi Benny,
thanks for your response and your hint!
> Hence, you should do connect call's media in on_call_media_state()
> callback instead.
Following this suggestion (as I understood it), I spent the past days shifting things around to achieve the desired result
- unfortunately, with no success.
Here is the condensed essence (proof-of-concept model) of the current implementation.
// some initialized globals ...
bool mergeCalls = false;
pjsua_call_id activeCallID = -1;
pjsua_call_id holdingCallID = -1;
// the GUI calls "mergeCallButtonPressed()", while "holdingCallID" had been "pjsua_call_set_hold(holdingCallID, NULL)" and "activeCallID" is actually talking:
void mergeCallButtonPressed()
{
mergeCalls = true;
pjsua_call_reinvite(holdingCallID, PJ_TRUE, NULL);
}
// pjsua_call_reinvite then triggers:
void on_call_media_state(pjsua_call_id callID)
{
pjsua_call_info callInfo;
pjsua_call_get_info(callID, &callInfo);
if (callInfo.media_status == PJSUA_CALL_MEDIA_ACTIVE) { // When media is active, connect call to sound device.
pjsua_conf_port_id slotOne = callInfo.conf_slot;
pjsua_conf_connect(slotOne, 0);
pjsua_conf_connect(0, slotOne);
if (mergeCalls == true) {
pjsua_conf_port_id slotTwo = pjsua_call_get_conf_port(activeCallID);
pjsua_conf_connect(slotOne, slotTwo);
pjsua_conf_connect(slotTwo, slotOne);
// since the "activeCallID" is already talking, its conf_port is already connected to "0" (and vice versa) ...
} else {
activeCallID = callID;
}
} else if (callInfo.media_status == PJSUA_CALL_MEDIA_LOCAL_HOLD) {
// ? callSuspended(callID);
}
}
While both remote-parties (activeCallID and holdingCallID) can listen and speak to port "0", they can neither listen nor speak to each other.
What am I missing/misunderstanding?
Again, thank you very much in advance for your help!
Best regards,
Thomas
On Oct 6, 2011, at 4:55 , Benny Prijono wrote:
> On Sat, Oct 1, 2011 at 5:46 PM, Thomas Martin <tmemail at gmx.de> wrote:
>> Hello Everybody,
>>
>> I am trying to implement a 3-party-conference by connecting the mediastreams as follows:
>>
>> void mergeCalls(int callOne, int callTwo) {
>>
>> // make sure that both calls are off-hold
>> pjsua_call_reinvite(callOne, PJ_TRUE, NULL);
>> pjsua_call_reinvite(callTwo, PJ_TRUE, NULL);
>>
>> // obtaining the media-slots
>> pjsua_conf_port_id slotOne = pjsua_call_get_conf_port(callOne);
>> pjsua_conf_port_id slotTwo = pjsua_call_get_conf_port(callTwo);
>>
>> // connecting everybody
>> pjsua_conf_connect(0, slotOne);
>> pjsua_conf_connect(slotOne, 0);
>>
>> pjsua_conf_connect(0, slotTwo);
>> pjsua_conf_connect(slotTwo, 0);
>>
>> pjsua_conf_connect(slotOne, slotTwo);
>> pjsua_conf_connect(slotTwo, slotOne);
>> }
>>
>
> The pjsua_conf_connect() above may not be useful, depending on what
> pjsua_call_reinvite() does. If the re-INVITE recreates the media
> stream (and most likely it does), your conf connection will be lost,
> so the pjsua_conf_connect() instructions above are useless.
>
> Hence, you should do connect call's media in on_call_media_state()
> callback instead.
>
> Benny
>
>
>
>> "mergeCalls()" is called, while two calls are connected and either callOne or callTwo is "holding" (as in "pjsua_call_set_hold(callID, NULL)").
>>
>> Even though the above code might be a bit of "overkill" (inefficient) with respect to re-inviting both calls and (re-)connecting both calls to "slot 0", this should not really be harmful.
>>
>> The Log states:
>>
>> 12:07:34.115 conference.c Port 3 (sip:12345 at provider.com;user=phone) transmitting to port 0 (iPhone IO device)
>> 12:07:34.115 conference.c Port 0 (iPhone IO device) transmitting to port 3 (sip:12345 at provider.com;user=phone)
>>
>> 12:07:37.677 conference.c Port 4 (sip:54321 at provider.com) transmitting to port 0 (iPhone IO device)
>> 12:07:37.677 conference.c Port 0 (iPhone IO device) transmitting to port 4 (sip:54321 at provider.com)
>>
>> 12:07:37.736 conference.c Port 4 (sip:54321 at provider.com) transmitting to port 3 (sip:12345 at provider.com;user=phone)
>> 12:07:37.736 conference.c Port 3 (sip:12345 at provider.com;user=phone) transmitting to port 4 (sip:54321 at provider.com)
>>
>> However, the mediastreams do not seem to get merged:
>>
>> Port 0 can listen and talk to both, Port 3 and Port 4
>> but:
>> Port 3 can listen and talk to Port 0 but NOT to Port 4
>> Port 4 can listen and talk to Port 0 but NOT to Port 3
>>
>> (Port 3 seems to be able to hear Port 4 for a fraction of a second)
>>
>> Somewhere, I had read about complications regarding re-pjsua_call_set_hold() and pjsua_call_reinvite() but I can't find the reference anymore.
>>
>> Any hints are greatly appreciated!
>>
>> Thank you very much in advance.
>>
>> Best Regards,
>>
>> Thomas
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
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>
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