No audio on answer about tg2sip HOT 10 OPEN

Can you elaborate more on that? Where, how, any hints/tips/steps would be helpful! :)

https://github.com/TelegramMessenger/Telegram-iOS/releases/tag/build-25451

hi i see some fixes with calls here maybe ios client is fixed now

you can try test it if you want)

We tested - no voice :(

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Also, it doesn't work with the pre-built binaries from here. Tried with universal 1.3.0

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@oksakhartman

• iOS (latest os and tg app version) - that's the first leg, I'm trying to telegram call it from tg2sip
• the second leg - Linphone (macOS, latest again), running on the same local network Asterisk
• calls are working fine on the iOS tg app - Tested by calling friends. Also few of the Linphone calls succeeds, and there was audio :)

Not sure if related, but

pjproject-git 2.13.r79.gf60d1c4-1 - built with

#define PJMEDIA_CONF_USE_SWITCH_BOARD 1
#define PJMEDIA_CONF_SWITCH_BOARD_BUF_SIZE 2000

I can paste the whole config_site.h if needed ...

Here's the configure command

./configure \
    --prefix=/usr \
    --enable-epoll \
    --with-external-speex \
    --with-external-srtp \
    --with-external-gsm \
    --disable-video \
    --disable-sound \
    --enable-shared

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@oksakhartman I've managed to test some scenarios :)

A - tg2sip (and Asterisk)
B - iOS, latest tg
C - iOS, linphone
D - Android, latest tg
E - Android, latest tg X
F - macos, latest tg client
G - macos, chrome web z
H - macos, linphone

H -> A -> B - no audio
B -> A -> H - no audio
H -> A -> D - works
H -> A -> E - works
C -> A -> H - works
C -> A -> G - can't answer, the web client says "You need to upgrade to latest version..." From there I've downloaded F (macos tg client)
C -> A -> F - works (audio connects after 2-3 seconds delay)
D -> A -> H - works
E -> A -> H - works
C -> A -> D - works
C -> A -> E - works
E -> B - no audio (emojis match)
D -> B - works
G -> B - works
F -> B - works

Not sure, why few of the calls had audio (also, 7-8 in a row), between H -> A -> B ? I'm starting to suspect, that if the call gets "routed" through "udp reflector" that "accepts" version 2.4.4 - it works :/

Here are the changes that I've made to get branch feature/libtgvoip-update building without errors

using tdlib commit id b3ab664a18f8611f4dfcd3054717504271eeaa7a (few commits after that, the lib apis changes and doesn't work anymore with the implementation here in tg2sip)

diff --git a/CMakeLists.txt b/CMakeLists.txt
index 8507dbe..9d76659 100644
--- a/CMakeLists.txt
+++ b/CMakeLists.txt
@@ -7,7 +7,7 @@ add_subdirectory(libtgvoip)
 
 find_package(PkgConfig REQUIRED)
 find_package(Threads REQUIRED)
-find_package(Td 1.7.10 REQUIRED)
+find_package(Td 1.8.0 REQUIRED)
 find_package(spdlog 0.17 REQUIRED)
 pkg_check_modules(PJSIP libpjproject>=2.8 REQUIRED)
 pkg_check_modules(OPUS opus REQUIRED)
@@ -57,4 +57,4 @@ target_include_directories(gen_db PRIVATE
         include)
 
 target_link_libraries(gen_db PRIVATE
-        Td::TdStatic)
\ No newline at end of file
+        Td::TdStatic)
diff --git a/tg2sip/queue.h b/tg2sip/queue.h
index bfe1fa0..fb98706 100755
--- a/tg2sip/queue.h
+++ b/tg2sip/queue.h
@@ -21,6 +21,7 @@
 #include <mutex>
 #include <queue>
 #include <condition_variable>
+#include <optional>
 
 template<typename T>
 class OptionalQueue {
diff --git a/tg2sip/sip.cpp b/tg2sip/sip.cpp
index 06672a6..8dccc9d 100755
--- a/tg2sip/sip.cpp
+++ b/tg2sip/sip.cpp
@@ -185,10 +185,10 @@ void Client::init_pj_endpoint(Settings &settings, LogWriter *sip_log_writer) {
     // and TG audio port clock rate so we MUST force
     // using 48kHz codecs for all SIP calls
     std::string codecId = settings.raw_pcm() ? "L16/48000/1" : "opus/48000/2";
-    CodecInfoVector codecVector = ep.codecEnum();
+    CodecInfoVector2 codecVector = ep.codecEnum2();
 
-    for (auto const &value : codecVector) {
-        ep.codecSetPriority(value->codecId, (pj_uint8_t) (value->codecId == codecId ? 255 : 0));
+    for (auto const value : codecVector) {
+        ep.codecSetPriority(value.codecId, (pj_uint8_t) (value.codecId == codecId ? 255 : 0));
     }
 
     TransportConfig t_cfg;
@@ -309,4 +309,4 @@ void Client::DialDtmf(pjsua_call_id call_id, const string &dtmf_digits) {
 
     auto call = it->second;
     call->dialDtmf(dtmf_digits);
-}
\ No newline at end of file
+}
diff --git a/tg2sip/utils.cpp b/tg2sip/utils.cpp
index c8da02b..60b5df9 100644
--- a/tg2sip/utils.cpp
+++ b/tg2sip/utils.cpp
@@ -15,7 +15,8 @@
  * along with this program; If not, see <https://www.gnu.org/licenses/>.
  */
 
+#include <regex>
 #include <algorithm>
 #include "utils.h"
 
-bool is_digits(const std::string &str) { return std::all_of(str.begin(), str.end(), ::isdigit); };
\ No newline at end of file
+bool is_digits(const std::string &str) { return std::regex_search(str, std::regex( "\\d+" )); };

from tg2sip.

@oksakhartman Yup, few calls had audio. Maybe 10 out of ~200 :) Also, one time it worked for 5-6 calls made in a row, then after restarting tg2sip doesn't worked anymore (still not worked from that time) :/

Looks like a bug, yeah ... If it was a "rejection" from deprecation, it shouldn't worked at all :)

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@oksakhartman

I switched to a fork on webrtc

Can you elaborate more on that? Where, how, any hints/tips/steps would be helpful! :)

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I switched to a fork on webrtc

I will be grateful if you provide compiled x64 binaries for tests

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We tested - no voice :(

https://github.com/TelegramMessenger/Telegram-iOS/releases/tag/build-25780

No voice :(

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No voice :(

вроде уже работает на айфонах

Что значит вроде, вы тестировали?

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No voice :(

вроде уже работает на айфонах

Протестировали, голоса нету.

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