hello all
I have special request. I want create call where RTP stream will be sent only
one way from caller to callee. Not from callee to caller.
exists some "method in SIP message" how can i say collee to not sent RTP
stream? Exists something like user data in SIP message? If yes how can i set
it?
how can i disable RTP stream on collee side? (some API) ..respectively on
UA side.I found callback on_stream_created() it is good place? :)
thanks
look at the sdp attributes
a=sendonly, rcvonly sendrcv
http://www.ietf.org/rfc/rfc3264.txt
Cheers
On Apr 13, 2010, at 10:25 AM, Peter Lukac wrote:
hello all
I have special request. I want create call where RTP stream will be sent only one way from caller to callee. Not from callee to caller.
exists some "method in SIP message" how can i say collee to not sent RTP stream? Exists something like user data in SIP message? If yes how can i set it?
how can i disable RTP stream on collee side? (some API) ..respectively on UA side.I found callback on_stream_created() it is good place? :)
thanks
Visit our blog: http://blog.pjsip.org
pjsip mailing list
pjsip@lists.pjsip.org
http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org
Alejandro Orellana
Savant Systems, LLC
770 Main Street
Osterville, Ma. 02655
(508) 683-2547 (W)
(508) 683-2647 (F)
Alejandro.Orellana@savantav.com
Statement of Confidentiality
The contents of this e-mail message and any attachments are confidential and are intended solely for the addressee. The information may also be legally privileged. This transmission is sent in trust, and the sole purpose of delivery to the intended recipient. If you have received this transmission in error, any use, reproduction or dissemination of this transmission is strictly prohibited. If you are not the intended recipient, please immediately notify the sender by reply e-mail or at 508.683-2500 and delete this message and its attachments, if any.
thanks for advice. I look at it.
look at the sdp attributes
a=sendonly, rcvonly sendrcv
http://www.ietf.org/rfc/rfc3264.txt
Cheers
On Apr 13, 2010, at 10:25 AM, Peter Lukac wrote:
hello all
I have special request. I want create call where RTP stream will be sent
only one way from caller to callee. Not from callee to caller.
exists some "method in SIP message" how can i say collee to not sent
RTP stream? Exists something like user data in SIP message? If yes how
can i set it?
how can i disable RTP stream on collee side? (some API) ..respectively
on UA side.I found callback on_stream_created() it is good place? :)
thanks
hello again
How can i add/modify DPS protocol in SIP message when i creating call?
in pjsua_call_make_call method is parameter msg_data..
pj_status_t pjsua_call_make_call
(
pjsua_acc_id
acc_id,
const pj_str_t *
dst_uri,
unsigned
options,
void *
user_data,
const pjsua_msg_data *
msg_data,
pjsua_call_id *
p_call_id
)
and i suppose this parameter i have to adjust. But how?
when i try:
pjsua_msg_data msg_data;
pjsua_msg_data_init( &msg_data );
pj_str_t hname = pj_str("application/sdp");
pj_str_t hvalue = pj_str("a=sendonly");
msg_data.content_type = hname;
msg_data.msg_body = hvalue;
pjsua_call_make_call( acc_id, &pj_uri, 0, NULL, &msg_data, &call_id);
in SIP message is still:
"a=sendrecv"
struct "pjsua_msg_data" have another member "hdr_list" of type "pjsip_hdr"
Do i have to adjust this member too?..if yes how?
thanks
Hi Peter,
On pjsua level, this may be easily done using this trick:
BR,
nanang
On Mon, Apr 26, 2010 at 7:23 PM, Peter Lukac p.lukac@emtest.sk wrote:
hello again
How can i add/modify DPS protocol in SIP message when i creating call?
in pjsua_call_make_call method is parameter msg_data..
pj_status_t pjsua_call_make_call (pjsua_acc_id acc_id,
const pj_str_t * dst_uri,
unsigned options,
void * user_data,
const pjsua_msg_data * msg_data,
pjsua_call_id * p_call_id
)
and i suppose this parameter i have to adjust. But how?
when i try:
pjsua_msg_data msg_data;
pjsua_msg_data_init( &msg_data );
pj_str_t hname = pj_str("application/sdp");
pj_str_t hvalue = pj_str("a=sendonly");
msg_data.content_type = hname;
msg_data.msg_body = hvalue;
pjsua_call_make_call( acc_id, &pj_uri, 0, NULL, &msg_data, &call_id);
in SIP message is still:
"a=sendrecv"
struct "pjsua_msg_data" have another member "hdr_list" of type "pjsip_hdr"
Do i have to adjust this member too?..if yes how?
thanks
Visit our blog: http://blog.pjsip.org
pjsip mailing list
pjsip@lists.pjsip.org
http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org
hi
thanks for reply.. this trick is little hack :) and i preffer clean
sollutions...it is posible with struct "pjsua_msg_data"? or some callback? or
some another way? ( use pjsua API is not necessary but welcome :) )
I want control rtp stream all directions (sendonly,recvonly and sendrecv).
thanks
Hi Peter,
On pjsua level, this may be easily done using this trick:
BR,
nanang
On Mon, Apr 26, 2010 at 7:23 PM, Peter Lukac p.lukac@emtest.sk wrote:
hello again
How can i add/modify DPS protocol in SIP message when i creating call?
in pjsua_call_make_call method is parameter msg_data..
pj_status_t pjsua_call_make_call (pjsua_acc_id acc_id,
const pj_str_t * dst_uri,
unsigned options,
void * user_data,
const pjsua_msg_data * msg_data,
pjsua_call_id * p_call_id
)
and i suppose this parameter i have to adjust. But how?
when i try:
pjsua_msg_data msg_data;
pjsua_msg_data_init( &msg_data );
pj_str_t hname = pj_str("application/sdp");
pj_str_t hvalue = pj_str("a=sendonly");
msg_data.content_type = hname;
msg_data.msg_body = hvalue;
pjsua_call_make_call( acc_id, &pj_uri, 0, NULL, &msg_data, &call_id);
in SIP message is still:
"a=sendrecv"
struct "pjsua_msg_data" have another member "hdr_list" of type
"pjsip_hdr" Do i have to adjust this member too?..if yes how?
thanks
Visit our blog: http://blog.pjsip.org
pjsip mailing list
pjsip@lists.pjsip.org
http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org
Visit our blog: http://blog.pjsip.org
pjsip mailing list
pjsip@lists.pjsip.org
http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org
hi
I found point where sdp message is creating.. is it in
endpoint.c source file and method "pjmedia_endpt_create_sdp"
here is code which add "sendrecv" attribute to SDP message:
/* Add sendrecv attribute. */
attr = PJ_POOL_ZALLOC_T(pool, pjmedia_sdp_attr);
attr->name = STR_SENDRECV;
m->attr[m->attr_count++] = attr;
as you see attribute "sendrecv" is still added. No condition if add
"sendonly", recvonly" or "sendrecv" attribute should be added :(
Maybe exist some callback which can overwrite? But i can't found anything...
..maybe call pjsua_call_set_hold() will be only solution :)
On Tuesday 27 April 2010 01:32:50 pm Peter Lukac wrote:
hi
thanks for reply.. this trick is little hack :) and i preffer clean
sollutions...it is posible with struct "pjsua_msg_data"? or some callback?
or some another way? ( use pjsua API is not necessary but welcome :) )
I want control rtp stream all directions (sendonly,recvonly and sendrecv).
thanks
Hi Peter,
On pjsua level, this may be easily done using this trick:
BR,
nanang
On Mon, Apr 26, 2010 at 7:23 PM, Peter Lukac p.lukac@emtest.sk wrote:
hello again
How can i add/modify DPS protocol in SIP message when i creating call?
in pjsua_call_make_call method is parameter msg_data..
pj_status_t pjsua_call_make_call (pjsua_acc_id acc_id,
const pj_str_t * dst_uri,
unsigned options,
void * user_data,
const pjsua_msg_data * msg_data,
pjsua_call_id * p_call_id
)
and i suppose this parameter i have to adjust. But how?
when i try:
pjsua_msg_data msg_data;
pjsua_msg_data_init( &msg_data );
pj_str_t hname = pj_str("application/sdp");
pj_str_t hvalue = pj_str("a=sendonly");
msg_data.content_type = hname;
msg_data.msg_body = hvalue;
pjsua_call_make_call( acc_id, &pj_uri, 0, NULL, &msg_data, &call_id);
in SIP message is still:
"a=sendrecv"
struct "pjsua_msg_data" have another member "hdr_list" of type
"pjsip_hdr" Do i have to adjust this member too?..if yes how?
thanks
Visit our blog: http://blog.pjsip.org
pjsip mailing list
pjsip@lists.pjsip.org
http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org
Visit our blog: http://blog.pjsip.org
pjsip mailing list
pjsip@lists.pjsip.org
http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org